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[alsa-devel] poll events?
by pl bossart 16 Dec '09

16 Dec '09

I am trying to understand what the expected behavior of poll() is in audio applications. I am using the pcm.c test app from alsa-lib with the following parameters. ./pcm -Dhw -c2 -r48000 -f440 --method write_and_poll The output looks like this: snd_pcm_dump(): Hardware PCM card 0 'Intel ICH6' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 16384 period_size : 8192 period_time : 170666 tstamp_mode : NONE period_step : 1 avail_min : 8192 period_event : 0 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 appl_ptr : 0 hw_ptr : 0 Now I instrumented the code in sound/core/pcm_native.c to show what events are returned by snd_pcm_playback_poll(), and I get the following pattern: [ 176.062986] ALSA: poll event POLLOUT|POLLWRNORM, avail 8193 avail_min 8192 [ 176.066630] ALSA: null poll event, avail 175 avail_min 8192 [ 176.233739] ALSA: poll event POLLOUT|POLLWRNORM, avail 8193 avail_min 8192 [ 176.237399] ALSA: null poll event, avail 176 avail_min 8192 [ 176.404503] ALSA: poll event POLLOUT|POLLWRNORM, avail 8193 avail_min 8192 [ 176.407652] ALSA: null poll event, avail 152 avail_min 8192 [ 176.575257] ALSA: poll event POLLOUT|POLLWRNORM, avail 8193 avail_min 8192 [ 176.578995] ALSA: null poll event, avail 180 avail_min 8192 [ 176.745999] ALSA: poll event POLLOUT|POLLWRNORM, avail 8193 avail_min 8192 [ 176.749663] ALSA: null poll event, avail 176 avail_min 8192 [ 176.916750] ALSA: poll event POLLOUT|POLLWRNORM, avail 8193 avail_min 8192 [ 176.919893] ALSA: null poll event, avail 151 avail_min 8192 It looks like I am getting additional wakeups from poll instead of waiting for a complete period. Is this supposed to be ok or is there something broken here? My expectation is that the hardware would only generate interrupts when a period has elapsed, i.e avail>=avail_min. The delta shown in dmesg between two POLLOUT events matches the period duration of 171 ms, I don't understand what generates these additional null events. Am I missing something? Thanks for your help. - Pierre

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[alsa-devel] Another 2.6.33 ASoC fix
by Mark Brown 15 Dec '09

15 Dec '09

The following changes since commit bc2580061e42c323d7777029f01318f395edac0d: Julia Lawall (1): ASoC: Correct code taking the size of a pointer are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Kuninori Morimoto (1): ASoC: ak4642: Add default return value in ak4642_modinit sound/soc/codecs/ak4642.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-)

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[alsa-devel] [PATCH v2] ASoC: ak4642: Add default return value in ak4642_modinit
by Kuninori Morimoto 15 Dec '09

15 Dec '09

If ak4642 driver was compiled without I2C configs, ak4642_modinit return value will become un-stable. This patch modify this bug Reported-by: Magnus Damm <damm(a)opensource.se> Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- v1 -> v2 o modify Magnus's email address So sorry ... sound/soc/codecs/ak4642.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index b69861d..3ef16bb 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -470,7 +470,7 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_ak4642); static int __init ak4642_modinit(void) { - int ret; + int ret = 0; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&ak4642_i2c_driver); #endif -- 1.6.3.3

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[alsa-devel] X-Fi Notebook express card
by The Source 15 Dec '09

15 Dec '09

Sorry for asking it here, but alsa-user mailing list is not responding. Has anyone tested yet if this card is supported? (not extreme audio, just X-Fi Notebook)

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Re: [alsa-devel] [PATCH fixed bug 4032 1/1] Fixed bug 4032.
by Takashi Iwai 15 Dec '09

15 Dec '09

At Mon, 14 Dec 2009 13:49:33 -0700, Steve Soule wrote: > > On 12/14/2009 12:01 PM, Takashi Iwai wrote: > > At Mon, 14 Dec 2009 11:41:55 -0700, > > Steve Soule wrote: > >> > >> >From c3387a767f827f57a6b1f6b772d3d7a1b6b39942 Mon Sep 17 00:00:00 2001 > >> From: Steve Soule <sts11dbxr(a)gmail.com> > >> Date: Mon, 14 Dec 2009 11:06:03 -0700 > >> Subject: [PATCH fixed bug 4032 1/1] Fixed bug 4032. > >> > >> > >> Signed-off-by: Steve Soule <sts11dbxr(a)gmail.com> > > > > Could you give a bit more changelog text? > > It's not sure what bug 4032 indicates, and more importantly, why this > > change is needed for fixing what. > > Okay. Here is a thorough changelog: > > This is a fix of a bug in ac97_codec.c. This bug has existed for many > years in many versions of alsa and versions of the kernel. The bug > appears with a Soundblaster 16PCI sound card. With this card, sometimes > the driver works correctly, and sometimes it doesn't. When it doesn't > work correctly, the driver prints the error message: > > AC'97 0 analog subsections not ready > > and fails to detect the complete set of sound controls. In particular, > the driver fails to detect the "Master Playback Volume" control, and so > the sound card volume level is arbitrary--usually very low. > > As far as I can tell, the problem is in the function snd_ac97_mixer() in > ac97_codec.c, in the lines immediately before the error message is > printed. These lines appear to be waiting for the card to come out of > powerdown mode. If the card does not come up within a timeout, the > error message is printed, and the controls are not detected correctly. > > The timeout in the latest version is 120 ms (though it was 100 ms when I > first reported this bug a year and a half ago). I experimented with > different timeout values, and found that 1 second was not sufficient, > and that 5 seconds is sufficient. Hrm, usually 1 second is really long enough for any operations. And 5 seconds is already a history since genesis. If it takes so long, something other is fishy. Although extending the timeout would be relatively harmless, I still don't think this is the point to fix. More likely the problem is in the codec accessor side... thanks, Takashi

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[alsa-devel] More HDA NID / control / proc related changes
by Jaroslav Kysela 15 Dec '09

15 Dec '09

Hi all, continuing the work of extending the HDA codec proc contents, I would like to introduce two new patches: ALSA: hda - add more NID->Control mapping ALSA: hda - introduce HDA_SUBDEV_AMP_FLAG (ControlAmp in proc) Patches can be obtained here: http://git.alsa-project.org/?p=alsa-kernel.git;a=shortlog;h=topic/hda-nid Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] [PATCH] ASoC: ak4642: Add default return value in ak4642_modinit
by Kuninori Morimoto 15 Dec '09

15 Dec '09

If ak4642 driver was compiled without I2C configs, ak4642_modinit return value will become un-stable. This patch modify this bug Reported-by: Magnus Damm <damm(a)igel.co.jp> Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- sound/soc/codecs/ak4642.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index b69861d..3ef16bb 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -470,7 +470,7 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_ak4642); static int __init ak4642_modinit(void) { - int ret; + int ret = 0; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&ak4642_i2c_driver); #endif -- 1.6.3.3

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[alsa-devel] Testing spam filter - please reply.
by Shareef Jalloq 15 Dec '09

15 Dec '09

Sorry - having amazing trouble trying to post to this mailing list. Replaced post with this text to see if it gets through. Shareef.

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[alsa-devel] [PATCH] sh: ms7724se: Add ak4642 support
by Kuninori Morimoto 15 Dec '09

15 Dec '09

Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- This patch is based on ALSA ak4642 patch. arch/sh/boards/mach-se/7724/setup.c | 17 +++++++++++++++++ 1 files changed, 17 insertions(+), 0 deletions(-) diff --git a/arch/sh/boards/mach-se/7724/setup.c b/arch/sh/boards/mach-se/7724/setup.c index 4b0f0c0..5d0f70b 100644 --- a/arch/sh/boards/mach-se/7724/setup.c +++ b/arch/sh/boards/mach-se/7724/setup.c @@ -514,6 +514,13 @@ static struct platform_device *ms7724se_devices[] __initdata = { &sdhi1_cn8_device, }; +/* I2C device */ +static struct i2c_board_info i2c0_devices[] = { + { + I2C_BOARD_INFO("ak4642", 0x12), + }, +}; + #define EEPROM_OP 0xBA206000 #define EEPROM_ADR 0xBA206004 #define EEPROM_DATA 0xBA20600C @@ -575,6 +582,16 @@ extern char ms7724se_sdram_enter_end; extern char ms7724se_sdram_leave_start; extern char ms7724se_sdram_leave_end; + +static int __init arch_setup(void) +{ + /* enable I2C device */ + i2c_register_board_info(0, i2c0_devices, + ARRAY_SIZE(i2c0_devices)); + return 0; +} +arch_initcall(arch_setup); + static int __init devices_setup(void) { u16 sw = ctrl_inw(SW4140); /* select camera, monitor */ -- 1.6.3.3

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[alsa-devel] Question about snd_soc_8_8_write
by Kuninori Morimoto 15 Dec '09

15 Dec '09

Dear Mark I have question about snd_soc_8_8_write ----------------------------- static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { u8 *cache = codec->reg_cache; u8 data[2]; BUG_ON(codec->volatile_register); data[0] = reg & 0xff; data[1] = value & 0xff; if (reg < codec->reg_cache_size) cache[reg] = value; if (codec->hw_write(codec->control_data, data, 2) == 2) return 0; else return -EIO; } ----------------------------- I wounder this behavior. if reg >= codec->reg_cache_size, this functions will call hw_write though cache[xx] ignore it. And even though if reg < codec->reg_cache_size, if hw_write return error, cache[xx] value was still changed. This is correct behavior ? Or are there any reason ? Best regards -- Kuninori Morimoto

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[alsa-devel] XMOS USB Audio Class 2.0 Reference Design
by Shareef Jalloq 15 Dec '09

15 Dec '09

Hello all, I'm starting a product development using the XMOS USB Audio Class 2.0 Ref Design (http://www.xmos.com/technology/prototype-boards/usbaudio2) and I'd like to get some help in understanding what support there is, or will be, for Audio Class 2.0 devices within Linux. First of all some disclaimers. I'm new to the Linux kernel and programming in general having come from a hardware background. I will do my best not to spam the mailing list but please don't assume something is too simple to explain to me. I'm going to use this experience as a learning exercise. I'm running Ubuntu 9.10 x64 : 2.6.31-16-generic. It is a fresh install so most things are default. I currently have the reference design hardware but am still waiting on XMOS to provide the device firmware. Once I have the source I should be able to answer any questions regarding desriptors in more detail. Until then I have the previously mentioned Ubuntu system and an OS X machine running 10.4.11. As far as I understand, 10.6.x is the only OS that currently supports the Audio Class 2.0 and I'm hoping to have access to a machine soon, but not yet. When I connect the device I get zero output. If I run lsub -v it is not recognised. If I check my syslog I get the following: Dec 10 00:16:00 sjalloq-desktop kernel: [53352.087740] usb 2-1.2.3: new high speed USB device using ehci_hcd and address 56 Dec 10 00:16:00 sjalloq-desktop kernel: [53352.107747] hub 2-1.2:1.0: unable to enumerate USB device on port 3 Dec 10 00:16:04 sjalloq-desktop kernel: [53356.286145] hub 2-1.2:1.0: Cannot enable port 3. Maybe the USB cable is bad? Dec 10 00:16:07 sjalloq-desktop kernel: [53359.285099] hub 2-1.2:1.0: Cannot enable port 3. Maybe the USB cable is bad? Dec 10 00:16:10 sjalloq-desktop kernel: [53362.284088] hub 2-1.2:1.0: Cannot enable port 3. Maybe the USB cable is bad? Dec 10 00:16:13 sjalloq-desktop kernel: [53365.282591] hub 2-1.2:1.0: Cannot enable port 3. Maybe the USB cable is bad? Dec 10 00:16:13 sjalloq-desktop kernel: [53365.282884] hub 2-1.2:1.0: unable to enumerate USB device on port 3 However, on my OS X machine, if I run USB Prober, I get the output that follows the end of the email. Is there anything else I can do that would be useful in debugging? Thanks, Shareef. __END__ OUTPUT FROM MAC OS X 10.4.11 USB PROBER: High Speed device @ 3 (0xFD100000): .............................. ............... Miscellaneous/Common Class device: "XMOS USB Audio 2.0" Device Descriptor Descriptor Version Number: 0x0200 Device Class: 239 (Miscellaneous) Device Subclass: 2 (Common Class) Device Protocol: 1 (Interface Association) Device MaxPacketSize: 64 Device VendorID/ProductID: 0x20B1/0x0002 (unknown vendor) Device Version Number: 0x0110 Number of Configurations: 1 Manufacturer String: 12 "XMOS " Product String: 7 "XMOS USB Audio 2.0" Serial Number String: 0 (none) Configuration Descriptor Length (and contents): 273 Raw Descriptor (hex) 0000: 09 02 11 01 03 01 00 80 FA 08 0B 00 03 01 00 20 Raw Descriptor (hex) 0010: 00 09 04 00 00 01 01 01 20 07 09 24 01 00 02 08 Raw Descriptor (hex) 0020: 7B 00 00 08 24 0A 28 03 07 00 01 11 24 02 02 01 Raw Descriptor (hex) 0030: 01 00 28 02 03 00 00 00 00 00 00 02 11 24 02 01 Raw Descriptor (hex) 0040: 03 06 00 28 04 0F 00 00 00 00 00 00 03 0C 24 03 Raw Descriptor (hex) 0050: 16 01 01 00 0B 28 00 00 04 0C 24 03 14 01 03 00 Raw Descriptor (hex) 0060: 0A 28 00 00 05 0C 24 03 16 01 03 00 0A 28 00 00 Raw Descriptor (hex) 0070: 06 12 24 06 0A 02 0F 00 00 00 0F 00 00 00 0F 00 Raw Descriptor (hex) 0080: 00 00 00 12 24 06 0B 01 0F 00 00 00 0F 00 00 00 Raw Descriptor (hex) 0090: 0F 00 00 00 00 07 05 83 03 06 00 08 09 04 01 00 Raw Descriptor (hex) 00a0: 00 01 02 20 08 09 04 01 01 02 01 02 20 09 10 24 Raw Descriptor (hex) 00b0: 01 02 00 01 01 00 00 00 02 03 00 00 00 00 06 24 Raw Descriptor (hex) 00c0: 02 01 04 18 07 05 01 05 00 04 01 08 25 01 00 00 Raw Descriptor (hex) 00d0: 02 08 00 07 05 81 11 04 00 08 09 04 02 00 00 01 Raw Descriptor (hex) 00e0: 02 20 0A 09 04 02 01 01 01 02 20 0B 10 24 01 16 Raw Descriptor (hex) 00f0: 00 01 01 00 00 00 02 03 00 00 00 00 06 24 02 01 Raw Descriptor (hex) 0100: 04 18 07 05 82 05 00 04 01 08 25 01 00 00 02 08 Raw Descriptor (hex) 0110: 00 Number of Interfaces: 3 Configuration Value: 1 Attributes: 0x80 (bus-powered) MaxPower: 500 ma Interface Association Audio/Unknown First Interface 0 Interface Count 3 Function Class 1 (Audio) Function Subclass 0 (Unknown) Interface Protocol 32 Function String 0 (none) Interface #0 - Audio/Control Alternate Setting 0 Number of Endpoints 1 Interface Class: 1 (Audio) Interface Subclass; 1 (Control) Interface Protocol: 32 Audio Control Class Specific Header Descriptor Version Number: 02.00 Category: 8 Class Specific Size: 123 bmControls: 0 Dump Contents (hex): 09 24 01 00 02 08 7B 00 00 Audio Class Specific Clock Source Clock ID: 40 Attributes: Internal Programmable Clock Controls: 7 Associated Terminal: 0 Terminal Name String Index: 1 Dump Contents (hex): 08 24 0A 28 03 07 00 01 Audio Class Specific Input Terminal Terminal ID: 2 Input Terminal Type: 0x101 (USB streaming) OutTerminal ID: 0 [NONE] Number of Channels: 2 Spatial config of channels: 0000000000000011 (null) (null) String index for first logical channel: 0 Terminal Name String Index: 2 Dump Contents (hex): 11 24 02 02 01 01 00 28 02 03 00 00 00 00 00 00 02 Audio Class Specific Input Terminal Terminal ID: 1 Input Terminal Type: 0x603 (Line connector) OutTerminal ID: 0 [NONE] Number of Channels: 4 Spatial config of channels: 0000000000001111 (null) (null) (null) (null) String index for first logical channel: 0 Terminal Name String Index: 3 Dump Contents (hex): 11 24 02 01 03 06 00 28 04 0F 00 00 00 00 00 00 03 Audio Class Specific Ouput Terminal Terminal ID: 22 Output Terminal Type: 0x101 (USB Isochronous Stream) InTerminal ID: 0 [NONE] Source ID: 11 Terminal Name String Index: 4 Dump Contents (hex): 0C 24 03 16 01 01 00 0B 28 00 00 04 Audio Class Specific Ouput Terminal Terminal ID: 20 Output Terminal Type: 0x301 (Speaker) InTerminal ID: 0 [NONE] Source ID: 10 Terminal Name String Index: 5 Dump Contents (hex): 0C 24 03 14 01 03 00 0A 28 00 00 05 Audio Class Specific Ouput Terminal Terminal ID: 22 Output Terminal Type: 0x301 (Speaker) InTerminal ID: 0 [NONE] Source ID: 10 Terminal Name String Index: 6 Dump Contents (hex): 0C 24 03 16 01 03 00 0A 28 00 00 06 Audio Class Specific Feature Unit ID: 10 Source ID: 2 Number of Channels (ch): 2 Master Channel:......................... ( Mute:RW Volume:RW ) Channel 1:............................. ( Mute:RW Volume:RW ) Channel 2:............................. ( Mute:RW Volume:RW ) Feature Unit Name String Index: 0 Dump Contents (hex): 12 24 06 0A 02 0F 00 00 00 0F 00 00 00 0F 00 00 00 00 Audio Class Specific Feature Unit ID: 11 Source ID: 1 Number of Channels (ch): 2 Master Channel:......................... ( Mute:RW Volume:RW ) Channel 1:............................. ( Mute:RW Volume:RW ) Channel 2:............................. ( Mute:RW Volume:RW ) Feature Unit Name String Index: 0 Dump Contents (hex): 12 24 06 0B 01 0F 00 00 00 0F 00 00 00 0F 00 00 00 00 Endpoint 0x83 - Interrupt Input Address: 0x83 (IN) Attributes: 0x03 (Interrupt no synchronization data endpoint) Max Packet Size: 6 (6 x 1 transactions opportunities per microframe) Polling Interval: 8 (128 microframes (16 msecs) ) Interface #1 - Audio/Streaming Alternate Setting 0 Number of Endpoints 0 Interface Class: 1 (Audio) Interface Subclass; 2 (Streaming) Interface Protocol: 32 Interface #1 - Audio/Streaming (#1) Alternate Setting 1 Number of Endpoints 2 Interface Class: 1 (Audio) Interface Subclass; 2 (Streaming) Interface Protocol: 32 Audio Control Class Specific Header Audio Stream General Endpoint Terminal ID: 2 Format Type: 1 Formats PCM Number of Channels 2 Channel Configuration 3 Dump Contents (hex): 10 24 01 02 00 01 01 00 00 00 02 03 00 00 00 00 Audio Class Specific Audio Data Format Audio Stream Format Type Desc. Format Type: 1 Slot Size: 4 Bit Resolution: 24 Dump Contents (hex): 06 24 02 01 04 18 Endpoint 0x01 - Isochronous Output Address: 0x01 (OUT) Attributes: 0x05 (Isochronous asynchronous data endpoint) Max Packet Size: 1024 (1024 x 1 transactions opportunities per microframe) Polling Interval: 1 (1 microframe (125 microsecs) ) Class-Specific AS Audio EndPoint - Control output Attributes: 0x00 bLockDelayUnits: 0x00 (UNDEFINED) wLockDelay: 2050 Endpoint 0x81 - Isochronous Input Address: 0x81 (IN) Attributes: 0x11 (Isochronous no synchronization feedback endpoint) Max Packet Size: 4 (4 x 1 transactions opportunities per microframe) Polling Interval: 8 (128 microframes (16 msecs) ) Interface #2 - Audio/Streaming Alternate Setting 0 Number of Endpoints 0 Interface Class: 1 (Audio) Interface Subclass; 2 (Streaming) Interface Protocol: 32 Interface #2 - Audio/Streaming (#1) Alternate Setting 1 Number of Endpoints 1 Interface Class: 1 (Audio) Interface Subclass; 2 (Streaming) Interface Protocol: 32 Audio Control Class Specific Header Audio Stream General Endpoint Terminal ID: 22 Format Type: 1 Formats PCM Number of Channels 2 Channel Configuration 3 Dump Contents (hex): 10 24 01 16 00 01 01 00 00 00 02 03 00 00 00 00 Audio Class Specific Audio Data Format Audio Stream Format Type Desc. Format Type: 1 Slot Size: 4 Bit Resolution: 24 Dump Contents (hex): 06 24 02 01 04 18 Endpoint 0x82 - Isochronous Input Address: 0x82 (IN) Attributes: 0x05 (Isochronous asynchronous data endpoint) Max Packet Size: 1024 (1024 x 1 transactions opportunities per microframe) Polling Interval: 1 (1 microframe (125 microsecs) ) Class-Specific AS Audio EndPoint - Control output Attributes: 0x00 bLockDelayUnits: 0x00 (UNDEFINED) wLockDelay: 2050 Device Qualifier Descriptor Descriptor Version Number: 0x0200 Device Class 239 (Miscellaneous) Device Subclass 2 (Common Class) Device Protocol 1 (Interface Association) Device MaxPacketSize: 64 Number of Configurations: 1 bReserved: 0 Other Speed Configuration Descriptor Length (and contents): 18 Raw Descriptor (hex) 0000: 09 07 12 00 01 00 00 C0 32 09 04 00 00 00 00 00 Raw Descriptor (hex) 0010: 00 00 Number of Interfaces: 1 Configuration Value: 0 Attributes: 0xC0 (self-powered) MaxPower: 100 ma Interface #0 - Unknown Alternate Setting 0 Number of Endpoints 0 Interface Class: 0 (Unknown) Interface Subclass; 0 Interface Protocol: 0

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[alsa-devel] ASoC update for 2.6.33
by Mark Brown 14 Dec '09

14 Dec '09

The following changes since commit 761c9d45d14e0afa3c0b8eb84b4075602e50533b: Olof Johansson (1): ASoC: Fix build of OMAP sound drivers are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Another small bugfix: Julia Lawall (1): ASoC: Correct code taking the size of a pointer sound/soc/codecs/wm8900.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-)

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[alsa-devel] [PATCH fixed bug 4032 1/1] Fixed bug 4032.
by Steve Soule 14 Dec '09

14 Dec '09

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[alsa-devel] [git pull] PCMCIA updates for 2.6.33, part 2
by Dominik Brodowski 14 Dec '09

14 Dec '09

Linus, two more PCMCIA updates for 2.6.33 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/brodo/pcmcia-2.6.git master Please pull from that location. The diffstat and list of changes is below, the two patches will be sent (at least) to the linux-pcmcia list. Thanks, Dominik ---- The following changes since commit f40542532e96dda5506eb76badea322f2ae4731c: Linus Torvalds (1): Merge branch 'ixp4xx' of git://git.kernel.org/.../chris/linux-2.6 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/brodo/pcmcia-2.6.git master Dominik Brodowski (2): pcmcia: remove unused IRQ_FIRST_SHARED pcmcia: CodingStyle fixes drivers/isdn/hardware/avm/avm_cs.c | 3 +- drivers/isdn/hisax/avma1_cs.c | 3 +- drivers/isdn/hisax/elsa_cs.c | 2 +- drivers/isdn/hisax/sedlbauer_cs.c | 2 +- drivers/isdn/hisax/teles_cs.c | 2 +- drivers/net/pcmcia/axnet_cs.c | 2 +- drivers/net/pcmcia/fmvj18x_cs.c | 2 +- drivers/net/pcmcia/pcnet_cs.c | 2 +- drivers/net/pcmcia/smc91c92_cs.c | 2 +- drivers/net/pcmcia/xirc2ps_cs.c | 2 +- drivers/pcmcia/Kconfig | 6 +- drivers/pcmcia/cardbus.c | 23 ++-- drivers/pcmcia/cistpl.c | 181 ++++++++++++-------- drivers/pcmcia/cs.c | 12 +- drivers/pcmcia/ds.c | 66 ++++---- drivers/pcmcia/pcmcia_ioctl.c | 45 +++--- drivers/pcmcia/pcmcia_resource.c | 37 +++-- drivers/pcmcia/rsrc_mgr.c | 14 +- drivers/pcmcia/rsrc_nonstatic.c | 325 ++++++++++++++++++----------------- drivers/pcmcia/socket_sysfs.c | 6 +- drivers/pcmcia/yenta_socket.c | 147 +++++++++++------ include/pcmcia/cs.h | 4 +- include/pcmcia/ds.h | 6 +- include/pcmcia/mem_op.h | 2 +- include/pcmcia/ss.h | 12 +- sound/pcmcia/pdaudiocf/pdaudiocf.c | 3 +- 26 files changed, 504 insertions(+), 407 deletions(-)

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[PATCH fixed bug 4032 1/1] Fixed bug 4032.
by Steve Soule 14 Dec '09

14 Dec '09

Signed-off-by: Steve Soule <sts11dbxr(a)gmail.com> --- pci/ac97/ac97_codec.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/pci/ac97/ac97_codec.c b/pci/ac97/ac97_codec.c index 20cb60a..c119206 100644 --- a/pci/ac97/ac97_codec.c +++ b/pci/ac97/ac97_codec.c @@ -2122,7 +2122,7 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, } /* nothing should be in powerdown mode */ snd_ac97_write_cache(ac97, AC97_GENERAL_PURPOSE, 0); - end_time = jiffies + msecs_to_jiffies(120); + end_time = jiffies + msecs_to_jiffies(5000); do { if ((snd_ac97_read(ac97, AC97_POWERDOWN) & 0x0f) == 0x0f) goto __ready_ok; -- 1.6.4.2

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[alsa-devel] [PATCH 1/2 v2] ASoc: Add DA7210 codec device support for ALSA
by Kuninori Morimoto 14 Dec '09

14 Dec '09

This original driver was created by Dialog Semiconductor, and cleanuped by Kuninori Morimoto. Special thanks to David Chen. This became very simple ASoC codec driver, and it is tested by EcoVec24 board. Signed-off-by: David Chen <Dajun.chen(a)diasemi.com> Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- v1 -> v2 o modify i2c read o remove DAPM / widgets o use fixed value for default sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/da7210.c | 586 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/da7210.h | 24 ++ 4 files changed, 616 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/da7210.c create mode 100644 sound/soc/codecs/da7210.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 52b005f..ff33b02 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -23,6 +23,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4671 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_MAX9877 if I2C + select SND_SOC_DA7210 if I2C select SND_SOC_PCM3008 select SND_SOC_SPDIF select SND_SOC_SSM2602 if I2C @@ -112,6 +113,9 @@ config SND_SOC_AK4671 config SND_SOC_CS4270 tristate +config SND_SOC_DA7210 + tristate + # Cirrus Logic CS4270 Codec VD = 3.3V Errata # Select if you are affected by the errata where the part will not function # if MCLK divide-by-1.5 is selected and VD is set to 3.3V. The driver will diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index dbaecb1..a495bf8 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -10,6 +10,7 @@ snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o snd-soc-cs4270-objs := cs4270.o snd-soc-cx20442-objs := cx20442.o +snd-soc-da7210-objs := da7210.o snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-spdif-objs := spdif_transciever.o @@ -66,6 +67,7 @@ obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o +obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c new file mode 100644 index 0000000..14f5f34 --- /dev/null +++ b/sound/soc/codecs/da7210.c @@ -0,0 +1,586 @@ +/* + * DA7210 ALSA Soc codec driver + * + * Copyright (c) 2009 Dialog Semiconductor + * Written by David Chen <Dajun.chen(a)diasemi.com> + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Cleanups by Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * Tested on SuperH Ecovec24 board with S16/S24 LE in 48KHz using I2S + * + * 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/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/initval.h> +#include <asm/div64.h> + +#include "da7210.h" + +/* DA7210 register space */ +#define DA7210_STATUS 0x02 +#define DA7210_STARTUP1 0x03 +#define DA7210_MIC_L 0x07 +#define DA7210_MIC_R 0x08 +#define DA7210_INMIX_L 0x0D +#define DA7210_INMIX_R 0x0E +#define DA7210_ADC_HPF 0x0F +#define DA7210_ADC 0x10 +#define DA7210_DAC_HPF 0x14 +#define DA7210_DAC_L 0x15 +#define DA7210_DAC_R 0x16 +#define DA7210_DAC_SEL 0x17 +#define DA7210_OUTMIX_L 0x1C +#define DA7210_OUTMIX_R 0x1D +#define DA7210_HP_L_VOL 0x21 +#define DA7210_HP_R_VOL 0x22 +#define DA7210_HP_CFG 0x23 +#define DA7210_DAI_SRC_SEL 0x25 +#define DA7210_DAI_CFG1 0x26 +#define DA7210_DAI_CFG3 0x28 +#define DA7210_PLL_DIV3 0x2B +#define DA7210_PLL 0x2C + +/* STARTUP1 bit fields */ +#define DA7210_SC_MST_EN (1 << 0) + +/* MIC_L bit fields */ +#define DA7210_MICBIAS_EN (1 << 6) +#define DA7210_MIC_L_EN (1 << 7) + +/* MIC_R bit fields */ +#define DA7210_MIC_R_EN (1 << 7) + +/* INMIX_L bit fields */ +#define DA7210_IN_L_EN (1 << 7) + +/* INMIX_R bit fields */ +#define DA7210_IN_R_EN (1 << 7) + +/* ADC_HPF bit fields */ +#define DA7210_ADC_VOICE_EN (1 << 7) + +/* ADC bit fields */ +#define DA7210_ADC_L_EN (1 << 3) +#define DA7210_ADC_R_EN (1 << 7) + +/* DAC_SEL bit fields */ +#define DA7210_DAC_L_SRC_DAI_L (4 << 0) +#define DA7210_DAC_L_EN (1 << 3) +#define DA7210_DAC_R_SRC_DAI_R (5 << 4) +#define DA7210_DAC_R_EN (1 << 7) + +/* OUTMIX_L bit fields */ +#define DA7210_OUT_L_EN (1 << 7) + +/* OUTMIX_R bit fields */ +#define DA7210_OUT_R_EN (1 << 7) + +/* HP_CFG bit fields */ +#define DA7210_HP_2CAP_MODE (1 << 1) +#define DA7210_HP_SENSE_EN (1 << 2) +#define DA7210_HP_L_EN (1 << 3) +#define DA7210_HP_MODE (1 << 6) +#define DA7210_HP_R_EN (1 << 7) + +/* DAI_SRC_SEL bit fields */ +#define DA7210_DAI_OUT_L_SRC (6 << 0) +#define DA7210_DAI_OUT_R_SRC (7 << 4) + +/* DAI_CFG1 bit fields */ +#define DA7210_DAI_WORD_S16_LE (0 << 0) +#define DA7210_DAI_WORD_S24_LE (2 << 0) +#define DA7210_DAI_FLEN_64BIT (1 << 2) +#define DA7210_DAI_MODE_MASTER (1 << 7) + +/* DAI_CFG3 bit fields */ +#define DA7210_DAI_FORMAT_I2SMODE (0 << 0) +#define DA7210_DAI_OE (1 << 3) +#define DA7210_DAI_EN (1 << 7) + +/*PLL_DIV3 bit fields */ +#define DA7210_MCLK_RANGE_10_20_MHZ (1 << 4) +#define DA7210_PLL_BYP (1 << 6) + +/* PLL bit fields */ +#define DA7210_PLL_FS_48000 (11 << 0) + +#define DA7210_VERSION "0.0.1" + +/* Codec private data */ +struct da7210_priv { + struct snd_soc_codec codec; +}; + +static struct snd_soc_codec *da7210_codec; + +/* + * Register cache + */ +static const u8 da7210_reg[] = { + 0x00, 0x11, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R0 - R7 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x08, /* R8 - RF */ + 0x00, 0x00, 0x00, 0x00, 0x08, 0x10, 0x10, 0x54, /* R10 - R17 */ + 0x40, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R18 - R1F */ + 0x00, 0x00, 0x00, 0x02, 0x00, 0x76, 0x00, 0x00, /* R20 - R27 */ + 0x04, 0x00, 0x00, 0x30, 0x2A, 0x00, 0x40, 0x00, /* R28 - R2F */ + 0x40, 0x00, 0x40, 0x00, 0x40, 0x00, 0x40, 0x00, /* R30 - R37 */ + 0x40, 0x00, 0x40, 0x00, 0x40, 0x00, 0x00, 0x00, /* R38 - R3F */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R40 - R4F */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R48 - R4F */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R50 - R57 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R58 - R5F */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R60 - R67 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R68 - R6F */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R70 - R77 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x54, 0x54, 0x00, /* R78 - R7F */ + 0x00, 0x00, 0x2C, 0x00, 0x00, 0x00, 0x00, 0x00, /* R80 - R87 */ + 0x00, /* R88 */ +}; + +/* + * Read da7210 register cache + */ +static inline u32 da7210_read_reg_cache(struct snd_soc_codec *codec, u32 reg) +{ + u8 *cache = codec->reg_cache; + BUG_ON(reg > ARRAY_SIZE(da7210_reg)); + return cache[reg]; +} + +/* + * Write to the da7210 register space + */ +static int da7210_write(struct snd_soc_codec *codec, u32 reg, u32 value) +{ + u8 *cache = codec->reg_cache; + u8 data[2]; + + BUG_ON(codec->volatile_register); + + data[0] = reg & 0xff; + data[1] = value & 0xff; + + if (reg >= codec->reg_cache_size) + return -EIO; + + if (2 != codec->hw_write(codec->control_data, data, 2)) + return -EIO; + + cache[reg] = value; + return 0; +} + +/* + * Read from the da7210 register space. + */ +static inline u32 da7210_read(struct snd_soc_codec *codec, u32 reg) +{ + if (DA7210_STATUS == reg) + return i2c_smbus_read_byte_data(codec->control_data, reg); + + return da7210_read_reg_cache(codec, reg); +} + +static int da7210_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_codec *codec = dai->codec; + + if (is_play) { + /* PlayBack Volume 40 */ + snd_soc_update_bits(codec, DA7210_HP_L_VOL, 0x3F, 40); + snd_soc_update_bits(codec, DA7210_HP_R_VOL, 0x3F, 40); + + /* Enable Out */ + snd_soc_update_bits(codec, DA7210_OUTMIX_L, 0x1F, 0x10); + snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10); + + } else { + /* Volume 7 */ + snd_soc_update_bits(codec, DA7210_MIC_L, 0x7, 0x7); + snd_soc_update_bits(codec, DA7210_MIC_R, 0x7, 0x7); + + /* Enable Mic */ + snd_soc_update_bits(codec, DA7210_INMIX_L, 0x1F, 0x1); + snd_soc_update_bits(codec, DA7210_INMIX_R, 0x1F, 0x1); + } + + return 0; +} + +/* + * Set PCM DAI word length. + */ +static int da7210_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + u32 dai_cfg1; + u32 reg, mask; + + /* set DAI source to Left and Right ADC */ + da7210_write(codec, DA7210_DAI_SRC_SEL, + DA7210_DAI_OUT_R_SRC | DA7210_DAI_OUT_L_SRC); + + /* Enable DAI */ + da7210_write(codec, DA7210_DAI_CFG3, DA7210_DAI_OE | DA7210_DAI_EN); + + dai_cfg1 = 0xFC & da7210_read(codec, DA7210_DAI_CFG1); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + dai_cfg1 |= DA7210_DAI_WORD_S16_LE; + break; + case SNDRV_PCM_FORMAT_S24_LE: + dai_cfg1 |= DA7210_DAI_WORD_S24_LE; + break; + default: + return -EINVAL; + } + + da7210_write(codec, DA7210_DAI_CFG1, dai_cfg1); + + /* FIXME + * + * It support 48K only now + */ + switch (params_rate(params)) { + case 48000: + if (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) { + reg = DA7210_DAC_HPF; + mask = DA7210_DAC_VOICE_EN; + } else { + reg = DA7210_ADC_HPF; + mask = DA7210_ADC_VOICE_EN; + } + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, reg, mask, 0); + + return 0; +} + +/* + * Set DAI mode and Format + */ +static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u32 dai_cfg1; + u32 dai_cfg3; + + dai_cfg1 = 0x7f & da7210_read(codec, DA7210_DAI_CFG1); + dai_cfg3 = 0xfc & da7210_read(codec, DA7210_DAI_CFG3); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + dai_cfg1 |= DA7210_DAI_MODE_MASTER; + break; + default: + return -EINVAL; + } + + /* FIXME + * + * It support I2S only now + */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + dai_cfg3 |= DA7210_DAI_FORMAT_I2SMODE; + break; + default: + return -EINVAL; + } + + /* FIXME + * + * It support 64bit data transmission only now + */ + dai_cfg1 |= DA7210_DAI_FLEN_64BIT; + + da7210_write(codec, DA7210_DAI_CFG1, dai_cfg1); + da7210_write(codec, DA7210_DAI_CFG3, dai_cfg3); + + return 0; +} + +#define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +/* DAI operations */ +static struct snd_soc_dai_ops da7210_dai_ops = { + .startup = da7210_startup, + .hw_params = da7210_hw_params, + .set_fmt = da7210_set_dai_fmt, +}; + +struct snd_soc_dai da7210_dai = { + .name = "DA7210 IIS", + .id = 0, + /* playback capabilities */ + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA7210_FORMATS, + }, + /* capture capabilities */ + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA7210_FORMATS, + }, + .ops = &da7210_dai_ops, +}; +EXPORT_SYMBOL_GPL(da7210_dai); + +/* + * Initialize the DA7210 driver + * register the mixer and dsp interfaces with the kernel + */ +static int da7210_init(struct da7210_priv *da7210) +{ + struct snd_soc_codec *codec = &da7210->codec; + int ret = 0; + + if (da7210_codec) { + dev_err(codec->dev, "Another da7210 is registered\n"); + return -EINVAL; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = da7210; + codec->name = "DA7210"; + codec->owner = THIS_MODULE; + codec->read = da7210_read; + codec->write = da7210_write; + codec->dai = &da7210_dai; + codec->num_dai = 1; + codec->hw_write = (hw_write_t)i2c_master_send; + codec->reg_cache_size = ARRAY_SIZE(da7210_reg); + codec->reg_cache = kmemdup(da7210_reg, + sizeof(da7210_reg), GFP_KERNEL); + + if (!codec->reg_cache) + return -ENOMEM; + + da7210_dai.dev = codec->dev; + da7210_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret) { + dev_err(codec->dev, "Failed to register CODEC: %d\n", ret); + goto init_err; + } + + ret = snd_soc_register_dai(&da7210_dai); + if (ret) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + goto init_err; + } + + /* FIXME + * + * This driver use fixed value here + */ + + /* + * ADC settings + */ + + /* Enable Left & Right MIC PGA and Mic Bias */ + da7210_write(codec, DA7210_MIC_L, DA7210_MIC_L_EN | DA7210_MICBIAS_EN); + da7210_write(codec, DA7210_MIC_R, DA7210_MIC_R_EN); + + /* Enable Left and Right input PGA */ + da7210_write(codec, DA7210_INMIX_L, DA7210_IN_L_EN); + da7210_write(codec, DA7210_INMIX_R, DA7210_IN_R_EN); + + /* Enable Left and Right ADC */ + da7210_write(codec, DA7210_ADC, DA7210_ADC_L_EN | DA7210_ADC_R_EN); + + /* + * DAC settings + */ + + /* Enable Left and Right DAC */ + da7210_write(codec, DA7210_DAC_SEL, + DA7210_DAC_L_SRC_DAI_L | DA7210_DAC_L_EN | + DA7210_DAC_R_SRC_DAI_R | DA7210_DAC_R_EN); + + /* Enable Left and Right out PGA */ + da7210_write(codec, DA7210_OUTMIX_L, DA7210_OUT_L_EN); + da7210_write(codec, DA7210_OUTMIX_R, DA7210_OUT_R_EN); + + /* Enable Left and Right HeadPhone PGA */ + da7210_write(codec, DA7210_HP_CFG, + DA7210_HP_2CAP_MODE | DA7210_HP_SENSE_EN | + DA7210_HP_L_EN | DA7210_HP_MODE | DA7210_HP_R_EN); + + /* Diable PLL and bypass it */ + da7210_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); + + /* Bypass PLL and set MCLK freq rang to 10-20MHz */ + da7210_write(codec, DA7210_PLL_DIV3, + DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); + + /* Activate all enabled subsystem */ + da7210_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); + + return ret; + +init_err: + kfree(codec->reg_cache); + codec->reg_cache = NULL; + + return ret; + +} + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static int da7210_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct da7210_priv *da7210; + struct snd_soc_codec *codec; + int ret; + + da7210 = kzalloc(sizeof(struct da7210_priv), GFP_KERNEL); + if (!da7210) + return -ENOMEM; + + codec = &da7210->codec; + codec->dev = &i2c->dev; + + i2c_set_clientdata(i2c, da7210); + codec->control_data = i2c; + + ret = da7210_init(da7210); + if (ret < 0) + pr_err("Failed to initialise da7210 audio codec\n"); + + return ret; +} + +static int da7210_i2c_remove(struct i2c_client *client) +{ + struct da7210_priv *da7210 = i2c_get_clientdata(client); + + snd_soc_unregister_dai(&da7210_dai); + kfree(da7210->codec.reg_cache); + kfree(da7210); + da7210_codec = NULL; + + return 0; +} + +static const struct i2c_device_id da7210_i2c_id[] = { + { "da7210", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, da7210_i2c_id); + +/* I2C codec control layer */ +static struct i2c_driver da7210_i2c_driver = { + .driver = { + .name = "DA7210 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = da7210_i2c_probe, + .remove = __devexit_p(da7210_i2c_remove), + .id_table = da7210_i2c_id, +}; +#endif + +static int da7210_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret; + + if (!da7210_codec) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = da7210_codec; + codec = da7210_codec; + + /* Register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) + goto pcm_err; + + dev_info(&pdev->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); + +pcm_err: + return ret; +} + +static int da7210_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_da7210 = { + .probe = da7210_probe, + .remove = da7210_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_da7210); + +static int __init da7210_modinit(void) +{ + int ret = 0; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&da7210_i2c_driver); +#endif + return ret; +} +module_init(da7210_modinit); + +static void __exit da7210_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&da7210_i2c_driver); +#endif +} +module_exit(da7210_exit); + +MODULE_DESCRIPTION("ASoC DA7210 driver"); +MODULE_AUTHOR("David Chen, Kuninori Morimoto"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/da7210.h b/sound/soc/codecs/da7210.h new file mode 100644 index 0000000..390d621 --- /dev/null +++ b/sound/soc/codecs/da7210.h @@ -0,0 +1,24 @@ +/* + * da7210.h -- audio driver for da7210 + * + * Copyright (c) 2009 Dialog Semiconductor + * Written by David Chen <Dajun.chen(a)diasemi.com> + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Cleanups by Kuninori Morimoto <morimoto.kuninori(a)renesas.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; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef _DA7210_H +#define _DA7210_H + +extern struct snd_soc_dai da7210_dai; +extern struct snd_soc_codec_device soc_codec_dev_da7210; + +#endif + -- 1.6.3.3

2 1

[alsa-devel] [PATCH 9/9] sound/soc/codecs: Correct code taking the size of a pointer
by Julia Lawall 14 Dec '09

14 Dec '09

From: Julia Lawall <julia(a)diku.dk> sizeof(codec->reg_cache) is just the size of the pointer. Elsewhere in the file, codec->reg_cache is used with sizeof(wm8900_reg_defaults), so the code is changed to do the same here. A simplified version of the semantic patch that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression *x; expression f; type T; @@ *f(...,(T)x,...) // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/soc/codecs/wm8900.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index c9438dd..dbc368c 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -199,7 +199,7 @@ static void wm8900_reset(struct snd_soc_codec *codec) snd_soc_write(codec, WM8900_REG_RESET, 0); memcpy(codec->reg_cache, wm8900_reg_defaults, - sizeof(codec->reg_cache)); + sizeof(wm8900_reg_defaults)); } static int wm8900_hp_event(struct snd_soc_dapm_widget *w,

3 3

[alsa-devel] Two codecs using the same I2S bus
by Niels Langendorff 14 Dec '09

14 Dec '09

Hi, for a samsung based project we have chosen two codecs, one for input, the other for output. The two codecs are using the same I2S bus due to audio requirements (HF audio in combination with EC) When having two machine implementations at the same time for the two codecs we will run into problems attaching to the I2S bus. A solution is to combine the machine implementation into one file, but then I probaly have to modify generic kernel code. I there some solution for this and/or example code Thanks in advance, Niels Langendorff

2 4

[alsa-devel] [PATCH] Recognise and use subdevice 0x3061 found in some HP dv6 notebooks.
by Kunal Gangakhedkar 14 Dec '09

14 Dec '09

Obvious patch. Recently bought an HP Pavilion dv6-1110ax notebook. The modparm 'model=hp-dv5' works, but I thought it was better if it were natively supported. Also, the 'hp_detect=1' hint (echo "hp_detect = 1" > /sys/class/sound/hwC0D0/hints) works nicely to mute the speakers when using the headphone jack(s). Is there any way to automatically enable this? I'm no alsa dev - just scratching my itch here ;) PS: I'm not subscribed to the list - so, please put me in Cc: in the replies. Thanks, Kunal Signed-off-by: Kunal Gangakhedkar <kunal.gangakhedkar(a)gmail.com> --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3d59f83..0ed9a09 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1725,6 +1725,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "HP HDX", STAC_HP_HDX), /* HDX16 */ SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3620, "HP dv6", STAC_HP_DV5), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3061, + "HP dv6", STAC_HP_DV5), /* HP Pavilion dv6-1110ax */ SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x7010, "HP", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233, -- 1.6.5.GIT

2 3

[alsa-devel] [PATCH] sb_mixer: convert pointer tables to mixer control tables
by Krzysztof Helt 14 Dec '09

14 Dec '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Convert table of pointers to mixer controls into tables of the mixer controls. It saves about 20% of the snd-sb-common module size reported by lsmod. The als4000 uses part of sb16's control table. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/sb/sb_mixer.c | 330 ++++++++++++++++------------------------------ 1 files changed, 115 insertions(+), 215 deletions(-) diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c index 318ff0c..8cfc41f 100644 --- a/sound/isa/sb/sb_mixer.c +++ b/sound/isa/sb/sb_mixer.c @@ -528,20 +528,11 @@ int snd_sbmixer_add_ctl(struct snd_sb *chip, const char *name, int index, int ty * SB 2.0 specific mixer elements */ -static struct sbmix_elem snd_sb20_ctl_master_play_vol = - SB_SINGLE("Master Playback Volume", SB_DSP20_MASTER_DEV, 1, 7); -static struct sbmix_elem snd_sb20_ctl_pcm_play_vol = - SB_SINGLE("PCM Playback Volume", SB_DSP20_PCM_DEV, 1, 3); -static struct sbmix_elem snd_sb20_ctl_synth_play_vol = - SB_SINGLE("Synth Playback Volume", SB_DSP20_FM_DEV, 1, 7); -static struct sbmix_elem snd_sb20_ctl_cd_play_vol = - SB_SINGLE("CD Playback Volume", SB_DSP20_CD_DEV, 1, 7); - -static struct sbmix_elem *snd_sb20_controls[] = { - &snd_sb20_ctl_master_play_vol, - &snd_sb20_ctl_pcm_play_vol, - &snd_sb20_ctl_synth_play_vol, - &snd_sb20_ctl_cd_play_vol +static struct sbmix_elem snd_sb20_controls[] = { + SB_SINGLE("Master Playback Volume", SB_DSP20_MASTER_DEV, 1, 7), + SB_SINGLE("PCM Playback Volume", SB_DSP20_PCM_DEV, 1, 3), + SB_SINGLE("Synth Playback Volume", SB_DSP20_FM_DEV, 1, 7), + SB_SINGLE("CD Playback Volume", SB_DSP20_CD_DEV, 1, 7) }; static unsigned char snd_sb20_init_values[][2] = { @@ -552,41 +543,24 @@ static unsigned char snd_sb20_init_values[][2] = { /* * SB Pro specific mixer elements */ -static struct sbmix_elem snd_sbpro_ctl_master_play_vol = - SB_DOUBLE("Master Playback Volume", SB_DSP_MASTER_DEV, SB_DSP_MASTER_DEV, 5, 1, 7); -static struct sbmix_elem snd_sbpro_ctl_pcm_play_vol = - SB_DOUBLE("PCM Playback Volume", SB_DSP_PCM_DEV, SB_DSP_PCM_DEV, 5, 1, 7); -static struct sbmix_elem snd_sbpro_ctl_pcm_play_filter = - SB_SINGLE("PCM Playback Filter", SB_DSP_PLAYBACK_FILT, 5, 1); -static struct sbmix_elem snd_sbpro_ctl_synth_play_vol = - SB_DOUBLE("Synth Playback Volume", SB_DSP_FM_DEV, SB_DSP_FM_DEV, 5, 1, 7); -static struct sbmix_elem snd_sbpro_ctl_cd_play_vol = - SB_DOUBLE("CD Playback Volume", SB_DSP_CD_DEV, SB_DSP_CD_DEV, 5, 1, 7); -static struct sbmix_elem snd_sbpro_ctl_line_play_vol = - SB_DOUBLE("Line Playback Volume", SB_DSP_LINE_DEV, SB_DSP_LINE_DEV, 5, 1, 7); -static struct sbmix_elem snd_sbpro_ctl_mic_play_vol = - SB_SINGLE("Mic Playback Volume", SB_DSP_MIC_DEV, 1, 3); -static struct sbmix_elem snd_sbpro_ctl_capture_source = +static struct sbmix_elem snd_sbpro_controls[] = { + SB_DOUBLE("Master Playback Volume", + SB_DSP_MASTER_DEV, SB_DSP_MASTER_DEV, 5, 1, 7), + SB_DOUBLE("PCM Playback Volume", + SB_DSP_PCM_DEV, SB_DSP_PCM_DEV, 5, 1, 7), + SB_SINGLE("PCM Playback Filter", SB_DSP_PLAYBACK_FILT, 5, 1), + SB_DOUBLE("Synth Playback Volume", + SB_DSP_FM_DEV, SB_DSP_FM_DEV, 5, 1, 7), + SB_DOUBLE("CD Playback Volume", SB_DSP_CD_DEV, SB_DSP_CD_DEV, 5, 1, 7), + SB_DOUBLE("Line Playback Volume", + SB_DSP_LINE_DEV, SB_DSP_LINE_DEV, 5, 1, 7), + SB_SINGLE("Mic Playback Volume", SB_DSP_MIC_DEV, 1, 3), { .name = "Capture Source", .type = SB_MIX_CAPTURE_PRO - }; -static struct sbmix_elem snd_sbpro_ctl_capture_filter = - SB_SINGLE("Capture Filter", SB_DSP_CAPTURE_FILT, 5, 1); -static struct sbmix_elem snd_sbpro_ctl_capture_low_filter = - SB_SINGLE("Capture Low-Pass Filter", SB_DSP_CAPTURE_FILT, 3, 1); - -static struct sbmix_elem *snd_sbpro_controls[] = { - &snd_sbpro_ctl_master_play_vol, - &snd_sbpro_ctl_pcm_play_vol, - &snd_sbpro_ctl_pcm_play_filter, - &snd_sbpro_ctl_synth_play_vol, - &snd_sbpro_ctl_cd_play_vol, - &snd_sbpro_ctl_line_play_vol, - &snd_sbpro_ctl_mic_play_vol, - &snd_sbpro_ctl_capture_source, - &snd_sbpro_ctl_capture_filter, - &snd_sbpro_ctl_capture_low_filter + }, + SB_SINGLE("Capture Filter", SB_DSP_CAPTURE_FILT, 5, 1), + SB_SINGLE("Capture Low-Pass Filter", SB_DSP_CAPTURE_FILT, 3, 1) }; static unsigned char snd_sbpro_init_values[][2] = { @@ -598,68 +572,42 @@ static unsigned char snd_sbpro_init_values[][2] = { /* * SB16 specific mixer elements */ -static struct sbmix_elem snd_sb16_ctl_master_play_vol = - SB_DOUBLE("Master Playback Volume", SB_DSP4_MASTER_DEV, (SB_DSP4_MASTER_DEV + 1), 3, 3, 31); -static struct sbmix_elem snd_sb16_ctl_3d_enhance_switch = - SB_SINGLE("3D Enhancement Switch", SB_DSP4_3DSE, 0, 1); -static struct sbmix_elem snd_sb16_ctl_tone_bass = - SB_DOUBLE("Tone Control - Bass", SB_DSP4_BASS_DEV, (SB_DSP4_BASS_DEV + 1), 4, 4, 15); -static struct sbmix_elem snd_sb16_ctl_tone_treble = - SB_DOUBLE("Tone Control - Treble", SB_DSP4_TREBLE_DEV, (SB_DSP4_TREBLE_DEV + 1), 4, 4, 15); -static struct sbmix_elem snd_sb16_ctl_pcm_play_vol = - SB_DOUBLE("PCM Playback Volume", SB_DSP4_PCM_DEV, (SB_DSP4_PCM_DEV + 1), 3, 3, 31); -static struct sbmix_elem snd_sb16_ctl_synth_capture_route = - SB16_INPUT_SW("Synth Capture Route", SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 6, 5); -static struct sbmix_elem snd_sb16_ctl_synth_play_vol = - SB_DOUBLE("Synth Playback Volume", SB_DSP4_SYNTH_DEV, (SB_DSP4_SYNTH_DEV + 1), 3, 3, 31); -static struct sbmix_elem snd_sb16_ctl_cd_capture_route = - SB16_INPUT_SW("CD Capture Route", SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 2, 1); -static struct sbmix_elem snd_sb16_ctl_cd_play_switch = - SB_DOUBLE("CD Playback Switch", SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 2, 1, 1); -static struct sbmix_elem snd_sb16_ctl_cd_play_vol = - SB_DOUBLE("CD Playback Volume", SB_DSP4_CD_DEV, (SB_DSP4_CD_DEV + 1), 3, 3, 31); -static struct sbmix_elem snd_sb16_ctl_line_capture_route = - SB16_INPUT_SW("Line Capture Route", SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 4, 3); -static struct sbmix_elem snd_sb16_ctl_line_play_switch = - SB_DOUBLE("Line Playback Switch", SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 4, 3, 1); -static struct sbmix_elem snd_sb16_ctl_line_play_vol = - SB_DOUBLE("Line Playback Volume", SB_DSP4_LINE_DEV, (SB_DSP4_LINE_DEV + 1), 3, 3, 31); -static struct sbmix_elem snd_sb16_ctl_mic_capture_route = - SB16_INPUT_SW("Mic Capture Route", SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 0, 0); -static struct sbmix_elem snd_sb16_ctl_mic_play_switch = - SB_SINGLE("Mic Playback Switch", SB_DSP4_OUTPUT_SW, 0, 1); -static struct sbmix_elem snd_sb16_ctl_mic_play_vol = - SB_SINGLE("Mic Playback Volume", SB_DSP4_MIC_DEV, 3, 31); -static struct sbmix_elem snd_sb16_ctl_pc_speaker_vol = - SB_SINGLE("Beep Volume", SB_DSP4_SPEAKER_DEV, 6, 3); -static struct sbmix_elem snd_sb16_ctl_capture_vol = - SB_DOUBLE("Capture Volume", SB_DSP4_IGAIN_DEV, (SB_DSP4_IGAIN_DEV + 1), 6, 6, 3); -static struct sbmix_elem snd_sb16_ctl_play_vol = - SB_DOUBLE("Playback Volume", SB_DSP4_OGAIN_DEV, (SB_DSP4_OGAIN_DEV + 1), 6, 6, 3); -static struct sbmix_elem snd_sb16_ctl_auto_mic_gain = - SB_SINGLE("Mic Auto Gain", SB_DSP4_MIC_AGC, 0, 1); - -static struct sbmix_elem *snd_sb16_controls[] = { - &snd_sb16_ctl_master_play_vol, - &snd_sb16_ctl_3d_enhance_switch, - &snd_sb16_ctl_tone_bass, - &snd_sb16_ctl_tone_treble, - &snd_sb16_ctl_pcm_play_vol, - &snd_sb16_ctl_synth_capture_route, - &snd_sb16_ctl_synth_play_vol, - &snd_sb16_ctl_cd_capture_route, - &snd_sb16_ctl_cd_play_switch, - &snd_sb16_ctl_cd_play_vol, - &snd_sb16_ctl_line_capture_route, - &snd_sb16_ctl_line_play_switch, - &snd_sb16_ctl_line_play_vol, - &snd_sb16_ctl_mic_capture_route, - &snd_sb16_ctl_mic_play_switch, - &snd_sb16_ctl_mic_play_vol, - &snd_sb16_ctl_pc_speaker_vol, - &snd_sb16_ctl_capture_vol, - &snd_sb16_ctl_play_vol, - &snd_sb16_ctl_auto_mic_gain +static struct sbmix_elem snd_sb16_controls[] = { + SB_DOUBLE("Master Playback Volume", + SB_DSP4_MASTER_DEV, (SB_DSP4_MASTER_DEV + 1), 3, 3, 31), + SB_DOUBLE("PCM Playback Volume", + SB_DSP4_PCM_DEV, (SB_DSP4_PCM_DEV + 1), 3, 3, 31), + SB16_INPUT_SW("Synth Capture Route", + SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 6, 5), + SB_DOUBLE("Synth Playback Volume", + SB_DSP4_SYNTH_DEV, (SB_DSP4_SYNTH_DEV + 1), 3, 3, 31), + SB16_INPUT_SW("CD Capture Route", + SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 2, 1), + SB_DOUBLE("CD Playback Switch", + SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 2, 1, 1), + SB_DOUBLE("CD Playback Volume", + SB_DSP4_CD_DEV, (SB_DSP4_CD_DEV + 1), 3, 3, 31), + SB16_INPUT_SW("Mic Capture Route", + SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 0, 0), + SB_SINGLE("Mic Playback Switch", SB_DSP4_OUTPUT_SW, 0, 1), + SB_SINGLE("Mic Playback Volume", SB_DSP4_MIC_DEV, 3, 31), + SB_SINGLE("Beep Volume", SB_DSP4_SPEAKER_DEV, 6, 3), + SB_DOUBLE("Capture Volume", + SB_DSP4_IGAIN_DEV, (SB_DSP4_IGAIN_DEV + 1), 6, 6, 3), + SB_DOUBLE("Playback Volume", + SB_DSP4_OGAIN_DEV, (SB_DSP4_OGAIN_DEV + 1), 6, 6, 3), + SB16_INPUT_SW("Line Capture Route", + SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 4, 3), + SB_DOUBLE("Line Playback Switch", + SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 4, 3, 1), + SB_DOUBLE("Line Playback Volume", + SB_DSP4_LINE_DEV, (SB_DSP4_LINE_DEV + 1), 3, 3, 31), + SB_SINGLE("Mic Auto Gain", SB_DSP4_MIC_AGC, 0, 1), + SB_SINGLE("3D Enhancement Switch", SB_DSP4_3DSE, 0, 1), + SB_DOUBLE("Tone Control - Bass", + SB_DSP4_BASS_DEV, (SB_DSP4_BASS_DEV + 1), 4, 4, 15), + SB_DOUBLE("Tone Control - Treble", + SB_DSP4_TREBLE_DEV, (SB_DSP4_TREBLE_DEV + 1), 4, 4, 15) }; static unsigned char snd_sb16_init_values[][2] = { @@ -678,46 +626,34 @@ static unsigned char snd_sb16_init_values[][2] = { /* * DT019x specific mixer elements */ -static struct sbmix_elem snd_dt019x_ctl_master_play_vol = - SB_DOUBLE("Master Playback Volume", SB_DT019X_MASTER_DEV, SB_DT019X_MASTER_DEV, 4,0, 15); -static struct sbmix_elem snd_dt019x_ctl_pcm_play_vol = - SB_DOUBLE("PCM Playback Volume", SB_DT019X_PCM_DEV, SB_DT019X_PCM_DEV, 4,0, 15); -static struct sbmix_elem snd_dt019x_ctl_synth_play_vol = - SB_DOUBLE("Synth Playback Volume", SB_DT019X_SYNTH_DEV, SB_DT019X_SYNTH_DEV, 4,0, 15); -static struct sbmix_elem snd_dt019x_ctl_cd_play_vol = - SB_DOUBLE("CD Playback Volume", SB_DT019X_CD_DEV, SB_DT019X_CD_DEV, 4,0, 15); -static struct sbmix_elem snd_dt019x_ctl_mic_play_vol = - SB_SINGLE("Mic Playback Volume", SB_DT019X_MIC_DEV, 4, 7); -static struct sbmix_elem snd_dt019x_ctl_pc_speaker_vol = - SB_SINGLE("Beep Volume", SB_DT019X_SPKR_DEV, 0, 7); -static struct sbmix_elem snd_dt019x_ctl_line_play_vol = - SB_DOUBLE("Line Playback Volume", SB_DT019X_LINE_DEV, SB_DT019X_LINE_DEV, 4,0, 15); -static struct sbmix_elem snd_dt019x_ctl_pcm_play_switch = - SB_DOUBLE("PCM Playback Switch", SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 2,1, 1); -static struct sbmix_elem snd_dt019x_ctl_synth_play_switch = - SB_DOUBLE("Synth Playback Switch", SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 4,3, 1); -static struct sbmix_elem snd_dt019x_ctl_capture_source = +static struct sbmix_elem snd_dt019x_controls[] = { + /* ALS4000 below has some parts which we might be lacking, + * e.g. snd_als4000_ctl_mono_playback_switch - check it! */ + SB_DOUBLE("Master Playback Volume", + SB_DT019X_MASTER_DEV, SB_DT019X_MASTER_DEV, 4, 0, 15), + SB_DOUBLE("PCM Playback Switch", + SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 2, 1, 1), + SB_DOUBLE("PCM Playback Volume", + SB_DT019X_PCM_DEV, SB_DT019X_PCM_DEV, 4, 0, 15), + SB_DOUBLE("Synth Playback Switch", + SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 4, 3, 1), + SB_DOUBLE("Synth Playback Volume", + SB_DT019X_SYNTH_DEV, SB_DT019X_SYNTH_DEV, 4, 0, 15), + SB_DOUBLE("CD Playback Switch", + SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 2, 1, 1), + SB_DOUBLE("CD Playback Volume", + SB_DT019X_CD_DEV, SB_DT019X_CD_DEV, 4, 0, 15), + SB_SINGLE("Mic Playback Switch", SB_DSP4_OUTPUT_SW, 0, 1), + SB_SINGLE("Mic Playback Volume", SB_DT019X_MIC_DEV, 4, 7), + SB_SINGLE("Beep Volume", SB_DT019X_SPKR_DEV, 0, 7), + SB_DOUBLE("Line Playback Switch", + SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 4, 3, 1), + SB_DOUBLE("Line Playback Volume", + SB_DT019X_LINE_DEV, SB_DT019X_LINE_DEV, 4, 0, 15), { .name = "Capture Source", .type = SB_MIX_CAPTURE_DT019X - }; - -static struct sbmix_elem *snd_dt019x_controls[] = { - /* ALS4000 below has some parts which we might be lacking, - * e.g. snd_als4000_ctl_mono_playback_switch - check it! */ - &snd_dt019x_ctl_master_play_vol, - &snd_dt019x_ctl_pcm_play_vol, - &snd_dt019x_ctl_synth_play_vol, - &snd_dt019x_ctl_cd_play_vol, - &snd_dt019x_ctl_mic_play_vol, - &snd_dt019x_ctl_pc_speaker_vol, - &snd_dt019x_ctl_line_play_vol, - &snd_sb16_ctl_mic_play_switch, - &snd_sb16_ctl_cd_play_switch, - &snd_sb16_ctl_line_play_switch, - &snd_dt019x_ctl_pcm_play_switch, - &snd_dt019x_ctl_synth_play_switch, - &snd_dt019x_ctl_capture_source + } }; static unsigned char snd_dt019x_init_values[][2] = { @@ -735,82 +671,37 @@ static unsigned char snd_dt019x_init_values[][2] = { /* * ALS4000 specific mixer elements */ -static struct sbmix_elem snd_als4000_ctl_master_mono_playback_switch = - SB_SINGLE("Master Mono Playback Switch", SB_ALS4000_MONO_IO_CTRL, 5, 1); -static struct sbmix_elem snd_als4k_ctl_master_mono_capture_route = { +static struct sbmix_elem snd_als4000_controls[] = { + SB_DOUBLE("PCM Playback Switch", + SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 2, 1, 1), + SB_DOUBLE("Synth Playback Switch", + SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 4, 3, 1), + SB_SINGLE("Mic Boost (+20dB)", SB_ALS4000_MIC_IN_GAIN, 0, 0x03), + SB_SINGLE("Master Mono Playback Switch", SB_ALS4000_MONO_IO_CTRL, 5, 1), + { .name = "Master Mono Capture Route", .type = SB_MIX_MONO_CAPTURE_ALS4K - }; -static struct sbmix_elem snd_als4000_ctl_mono_playback_switch = - SB_SINGLE("Mono Playback Switch", SB_DT019X_OUTPUT_SW2, 0, 1); -static struct sbmix_elem snd_als4000_ctl_mic_20db_boost = - SB_SINGLE("Mic Boost (+20dB)", SB_ALS4000_MIC_IN_GAIN, 0, 0x03); -static struct sbmix_elem snd_als4000_ctl_mixer_analog_loopback = - SB_SINGLE("Analog Loopback Switch", SB_ALS4000_MIC_IN_GAIN, 7, 0x01); -static struct sbmix_elem snd_als4000_ctl_mixer_digital_loopback = + }, + SB_SINGLE("Mono Playback Switch", SB_DT019X_OUTPUT_SW2, 0, 1), + SB_SINGLE("Analog Loopback Switch", SB_ALS4000_MIC_IN_GAIN, 7, 0x01), + SB_SINGLE("3D Control - Switch", SB_ALS4000_3D_SND_FX, 6, 0x01), SB_SINGLE("Digital Loopback Switch", - SB_ALS4000_CR3_CONFIGURATION, 7, 0x01); -/* FIXME: functionality of 3D controls might be swapped, I didn't find - * a description of how to identify what is supposed to be what */ -static struct sbmix_elem snd_als4000_3d_control_switch = - SB_SINGLE("3D Control - Switch", SB_ALS4000_3D_SND_FX, 6, 0x01); -static struct sbmix_elem snd_als4000_3d_control_ratio = - SB_SINGLE("3D Control - Level", SB_ALS4000_3D_SND_FX, 0, 0x07); -static struct sbmix_elem snd_als4000_3d_control_freq = + SB_ALS4000_CR3_CONFIGURATION, 7, 0x01), + /* FIXME: functionality of 3D controls might be swapped, I didn't find + * a description of how to identify what is supposed to be what */ + SB_SINGLE("3D Control - Level", SB_ALS4000_3D_SND_FX, 0, 0x07), /* FIXME: maybe there's actually some standard 3D ctrl name for it?? */ - SB_SINGLE("3D Control - Freq", SB_ALS4000_3D_SND_FX, 4, 0x03); -static struct sbmix_elem snd_als4000_3d_control_delay = + SB_SINGLE("3D Control - Freq", SB_ALS4000_3D_SND_FX, 4, 0x03), /* FIXME: ALS4000a.pdf mentions BBD (Bucket Brigade Device) time delay, * but what ALSA 3D attribute is that actually? "Center", "Depth", * "Wide" or "Space" or even "Level"? Assuming "Wide" for now... */ - SB_SINGLE("3D Control - Wide", SB_ALS4000_3D_TIME_DELAY, 0, 0x0f); -static struct sbmix_elem snd_als4000_3d_control_poweroff_switch = - SB_SINGLE("3D PowerOff Switch", SB_ALS4000_3D_TIME_DELAY, 4, 0x01); -static struct sbmix_elem snd_als4000_ctl_3db_freq_control_switch = + SB_SINGLE("3D Control - Wide", SB_ALS4000_3D_TIME_DELAY, 0, 0x0f), + SB_SINGLE("3D PowerOff Switch", SB_ALS4000_3D_TIME_DELAY, 4, 0x01), SB_SINGLE("Master Playback 8kHz / 20kHz LPF Switch", - SB_ALS4000_FMDAC, 5, 0x01); + SB_ALS4000_FMDAC, 5, 0x01), #ifdef NOT_AVAILABLE -static struct sbmix_elem snd_als4000_ctl_fmdac = - SB_SINGLE("FMDAC Switch (Option ?)", SB_ALS4000_FMDAC, 0, 0x01); -static struct sbmix_elem snd_als4000_ctl_qsound = - SB_SINGLE("QSound Mode", SB_ALS4000_QSOUND, 1, 0x1f); -#endif - -static struct sbmix_elem *snd_als4000_controls[] = { - /* ALS4000a.PDF regs page */ - &snd_sb16_ctl_master_play_vol, /* MX30/31 12 */ - &snd_dt019x_ctl_pcm_play_switch, /* MX4C 16 */ - &snd_sb16_ctl_pcm_play_vol, /* MX32/33 12 */ - &snd_sb16_ctl_synth_capture_route, /* MX3D/3E 14 */ - &snd_dt019x_ctl_synth_play_switch, /* MX4C 16 */ - &snd_sb16_ctl_synth_play_vol, /* MX34/35 12/13 */ - &snd_sb16_ctl_cd_capture_route, /* MX3D/3E 14 */ - &snd_sb16_ctl_cd_play_switch, /* MX3C 14 */ - &snd_sb16_ctl_cd_play_vol, /* MX36/37 13 */ - &snd_sb16_ctl_line_capture_route, /* MX3D/3E 14 */ - &snd_sb16_ctl_line_play_switch, /* MX3C 14 */ - &snd_sb16_ctl_line_play_vol, /* MX38/39 13 */ - &snd_sb16_ctl_mic_capture_route, /* MX3D/3E 14 */ - &snd_als4000_ctl_mic_20db_boost, /* MX4D 16 */ - &snd_sb16_ctl_mic_play_switch, /* MX3C 14 */ - &snd_sb16_ctl_mic_play_vol, /* MX3A 13 */ - &snd_sb16_ctl_pc_speaker_vol, /* MX3B 14 */ - &snd_sb16_ctl_capture_vol, /* MX3F/40 15 */ - &snd_sb16_ctl_play_vol, /* MX41/42 15 */ - &snd_als4000_ctl_master_mono_playback_switch, /* MX4C 16 */ - &snd_als4k_ctl_master_mono_capture_route, /* MX4B 16 */ - &snd_als4000_ctl_mono_playback_switch, /* MX4C 16 */ - &snd_als4000_ctl_mixer_analog_loopback, /* MX4D 16 */ - &snd_als4000_ctl_mixer_digital_loopback, /* CR3 21 */ - &snd_als4000_3d_control_switch, /* MX50 17 */ - &snd_als4000_3d_control_ratio, /* MX50 17 */ - &snd_als4000_3d_control_freq, /* MX50 17 */ - &snd_als4000_3d_control_delay, /* MX51 18 */ - &snd_als4000_3d_control_poweroff_switch, /* MX51 18 */ - &snd_als4000_ctl_3db_freq_control_switch, /* MX4F 17 */ -#ifdef NOT_AVAILABLE - &snd_als4000_ctl_fmdac, - &snd_als4000_ctl_qsound, + SB_SINGLE("FMDAC Switch (Option ?)", SB_ALS4000_FMDAC, 0, 0x01), + SB_SINGLE("QSound Mode", SB_ALS4000_QSOUND, 1, 0x1f), #endif }; @@ -829,11 +720,10 @@ static unsigned char snd_als4000_init_values[][2] = { { SB_ALS4000_MIC_IN_GAIN, 0 }, }; - /* */ static int snd_sbmixer_init(struct snd_sb *chip, - struct sbmix_elem **controls, + struct sbmix_elem *controls, int controls_count, unsigned char map[][2], int map_count, @@ -856,7 +746,8 @@ static int snd_sbmixer_init(struct snd_sb *chip, } for (idx = 0; idx < controls_count; idx++) { - if ((err = snd_sbmixer_add_ctl_elem(chip, controls[idx])) < 0) + err = snd_sbmixer_add_ctl_elem(chip, &controls[idx]); + if (err < 0) return err; } snd_component_add(card, name); @@ -908,6 +799,15 @@ int snd_sbmixer_new(struct snd_sb *chip) return err; break; case SB_HW_ALS4000: + /* use only the first 16 controls from SB16 */ + err = snd_sbmixer_init(chip, + snd_sb16_controls, + 16, + snd_sb16_init_values, + ARRAY_SIZE(snd_sb16_init_values), + "ALS4000"); + if (err < 0) + return err; if ((err = snd_sbmixer_init(chip, snd_als4000_controls, ARRAY_SIZE(snd_als4000_controls), -- 1.6.4 ---------------------------------------------------------------------- Wybierz samochod roku i wygraj! Sprawdz >>> http://link.interia.pl/f24e6

2 1

[alsa-devel] [PATCH] ALSA: hda: Fix max PCM level to 0 dB for AD1981_HP
by Dan Chen 14 Dec '09

14 Dec '09

Hi, Takashi, Attached is a patch against sound-2.6 (master) HEAD capping PCM at 0 dB for the AD1981_HP model. Thanks, Dan

2 1

[alsa-devel] [PATCH] ALSA: hda: Use ALC260_WILL quirk for another Acer model (0x1025007f)
by Dan Chen 14 Dec '09

14 Dec '09

Hi, Takashi, Here's a patch against sound-2.6 (master) HEAD to use the appropriate quirk for another Acer model. Thanks, Dan

2 1

[alsa-devel] [PATCH 8/9] sound/isa/gus: Correct code taking the size of a pointer
by Julia Lawall 14 Dec '09

14 Dec '09

From: Julia Lawall <julia(a)diku.dk> sizeof(share_id) is just the size of the pointer. On the other hand, block->share_id is an array, so its size seems more appropriate. A simplified version of the semantic patch that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression *x; expression f; type T; @@ *f(...,(T)x,...) // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/isa/gus/gus_mem.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/isa/gus/gus_mem.c b/sound/isa/gus/gus_mem.c index 661205c..af888a0 100644 --- a/sound/isa/gus/gus_mem.c +++ b/sound/isa/gus/gus_mem.c @@ -127,7 +127,8 @@ static struct snd_gf1_mem_block *snd_gf1_mem_share(struct snd_gf1_mem * alloc, !share_id[2] && !share_id[3]) return NULL; for (block = alloc->first; block; block = block->next) - if (!memcmp(share_id, block->share_id, sizeof(share_id))) + if (!memcmp(share_id, block->share_id, + sizeof(block->share_id))) return block; return NULL; }

2 1

[alsa-devel] [PATCH 2/2 v2] ASoC: Add FSI-DA7210 sound support for SuperH
by Kuninori Morimoto 14 Dec '09

14 Dec '09

Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- v1 -> v2 o remove strange return o remove add_i2c_device function sound/soc/sh/Kconfig | 8 ++++ sound/soc/sh/Makefile | 2 + sound/soc/sh/fsi-da7210.c | 83 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 93 insertions(+), 0 deletions(-) create mode 100644 sound/soc/sh/fsi-da7210.c diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 9e69765..8072a6d 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -47,4 +47,12 @@ config SND_FSI_AK4642 This option enables generic sound support for the FSI - AK4642 unit +config SND_FSI_DA7210 + bool "FSI-DA7210 sound support" + depends on SND_SOC_SH4_FSI + select SND_SOC_DA7210 + help + This option enables generic sound support for the + FSI - DA7210 unit + endmenu diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile index a699787..1d0ec0a 100644 --- a/sound/soc/sh/Makefile +++ b/sound/soc/sh/Makefile @@ -13,6 +13,8 @@ obj-$(CONFIG_SND_SOC_SH4_FSI) += snd-soc-fsi.o ## boards snd-soc-sh7760-ac97-objs := sh7760-ac97.o snd-soc-fsi-ak4642-objs := fsi-ak4642.o +snd-soc-fsi-da7210-objs := fsi-da7210.o obj-$(CONFIG_SND_SH7760_AC97) += snd-soc-sh7760-ac97.o obj-$(CONFIG_SND_FSI_AK4642) += snd-soc-fsi-ak4642.o +obj-$(CONFIG_SND_FSI_DA7210) += snd-soc-fsi-da7210.o diff --git a/sound/soc/sh/fsi-da7210.c b/sound/soc/sh/fsi-da7210.c new file mode 100644 index 0000000..33b4d17 --- /dev/null +++ b/sound/soc/sh/fsi-da7210.c @@ -0,0 +1,83 @@ +/* + * fsi-da7210.c + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.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; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/io.h> +#include <linux/i2c.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <sound/sh_fsi.h> +#include "../codecs/da7210.h" + +static int fsi_da7210_init(struct snd_soc_codec *codec) +{ + return snd_soc_dai_set_fmt(&da7210_dai, + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); +} + +static struct snd_soc_dai_link fsi_da7210_dai = { + .name = "DA7210", + .stream_name = "DA7210", + .cpu_dai = &fsi_soc_dai[1], /* FSI B */ + .codec_dai = &da7210_dai, + .init = fsi_da7210_init, +}; + +static struct snd_soc_card fsi_soc_card = { + .name = "FSI", + .platform = &fsi_soc_platform, + .dai_link = &fsi_da7210_dai, + .num_links = 1, +}; + +static struct snd_soc_device fsi_da7210_snd_devdata = { + .card = &fsi_soc_card, + .codec_dev = &soc_codec_dev_da7210, +}; + +static struct platform_device *fsi_da7210_snd_device; + +static int __init fsi_da7210_sound_init(void) +{ + int ret; + + fsi_da7210_snd_device = platform_device_alloc("soc-audio", -1); + if (!fsi_da7210_snd_device) + return -ENOMEM; + + platform_set_drvdata(fsi_da7210_snd_device, &fsi_da7210_snd_devdata); + fsi_da7210_snd_devdata.dev = &fsi_da7210_snd_device->dev; + ret = platform_device_add(fsi_da7210_snd_device); + if (ret) + platform_device_put(fsi_da7210_snd_device); + + return ret; +} + +static void __exit fsi_da7210_sound_exit(void) +{ + platform_device_unregister(fsi_da7210_snd_device); +} + +module_init(fsi_da7210_sound_init); +module_exit(fsi_da7210_sound_exit); + +/* Module information */ +MODULE_DESCRIPTION("ALSA SoC FSI DA2710"); +MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori(a)renesas.com>"); +MODULE_LICENSE("GPL"); -- 1.6.3.3

1 0

Re: [alsa-devel] Q: Why I do have to restart alsa everytime after a resume from disk/ram ? Is this alsa- or kernel or config related??
by Robert Hancock 13 Dec '09

13 Dec '09

On 12/11/2009 03:21 AM, Christian Hartmann wrote: > hello list, > > everytime I suspended to disk or to ram, regardless of the used kernel > (git or stable from distri), I must restart alsa to hear sound again. > searched google and tried a lot but still the same awful behaviour. > Has anybode some suggestions for me? is this alsa related ? or is this > a kernel issue which could be dealt here? I'm not sure what you mean by restarting ALSA, do you mean reloading the audio device module or something? You didn't mention which kind of sound hardware you were using either..

3 3

[alsa-devel] [PATCH] Overwrite pin config on intel DG45ID board.
by Alexey Fisher 12 Dec '09

12 Dec '09

The pin config provided by BIOS have some problems: 0x0221401f: [Jack] HP Out at Ext Front <-- other association and sequence 0x02a19020: [Jack] Mic at Ext Front <-- other association 0x01113014: [Jack] Speaker at Ext Rear <-- line out (not speaker) 0x01114010: [Jack] Speaker at Ext Rear <-- line out 0x01a19030: [Jack] Mic at Ext Rear <-- other association 0x01111012: [Jack] Speaker at Ext Rear <-- line out 0x01116011: [Jack] Speaker at Ext Rear <-- line out 0x40f000f0: [N/A] Other at Ext N/A 0x40f000f0: [N/A] Other at Ext N/A 0x40f000f0: [N/A] Other at Ext N/A 0x40f000f0: [N/A] Other at Ext N/A 0x40f000f0: [N/A] Other at Ext N/A 0x01451140: [Jack] SPDIF Out at Ext Rear 0x40f000f0: [N/A] Other at Ext N/A just overwrite it. Signed-off-by: Alexey Fisher <bug-track(a)fisher-privat.net> --- sound/pci/hda/patch_sigmatel.c | 8 ++++++++ 1 files changed, 8 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 6b0bc04..395c174 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1538,6 +1538,13 @@ static unsigned int alienware_m17x_pin_configs[13] = { 0x904601b0, }; +static unsigned int intel_dg45id_pin_configs[14] = { + 0x02214230, 0x02A19240, 0x01013214, 0x01014210, + 0x01A19250, 0x01011212, 0x01016211, 0x40f000f0, + 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x014510A0, + 0x074510B0, 0x40f000f0 +}; + static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = { [STAC_92HD73XX_REF] = ref92hd73xx_pin_configs, [STAC_DELL_M6_AMIC] = dell_m6_pin_configs, @@ -1545,6 +1552,7 @@ static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = { [STAC_DELL_M6_BOTH] = dell_m6_pin_configs, [STAC_DELL_EQ] = dell_m6_pin_configs, [STAC_ALIENWARE_M17X] = alienware_m17x_pin_configs, + [STAC_92HD73XX_INTEL] = intel_dg45id_pin_configs, }; static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = { -- 1.6.3.3

2 1

[alsa-devel] [PATCH] opti93x: use dB scale for mixer controls
by Krzysztof Helt 12 Dec '09

12 Dec '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Add dB scale for mixer controls. Fix dB scale for Master Volume control. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The opti93x control patches can wait till next Linux version. I would like to see the patch "opti93x: fix irq releasing if the irq cannot be allocated" in this release as this is a typical bug fix. Regards, Krzysztof sound/isa/opti9xx/opti92x-ad1848.c | 36 ++++++++++++++++++++++-------------- 1 files changed, 22 insertions(+), 14 deletions(-) diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 3f55b31..a4af53b 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -547,32 +547,40 @@ __skip_mpu: #ifdef OPTi93X -static const DECLARE_TLV_DB_SCALE(db_scale_6bit, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_5bit_3db_step, -9300, 300, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_5bit, -4650, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_4bit_12db_max, -3300, 300, 0); static struct snd_kcontrol_new snd_opti93x_controls[] = { WSS_DOUBLE("Master Playback Switch", 0, OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1), WSS_DOUBLE_TLV("Master Playback Volume", 0, OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1, - db_scale_6bit), -WSS_DOUBLE("PCM Playback Volume", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 31, 1), -WSS_DOUBLE("FM Playback Volume", 0, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 1, 1, 15, 1), + db_scale_5bit_3db_step), +WSS_DOUBLE_TLV("PCM Playback Volume", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 31, 1, + db_scale_5bit), +WSS_DOUBLE_TLV("FM Playback Volume", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 1, 1, 15, 1, + db_scale_4bit_12db_max), WSS_DOUBLE("Line Playback Switch", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), -WSS_DOUBLE("Line Playback Volume", 0, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 15, 1), +WSS_DOUBLE_TLV("Line Playback Volume", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 15, 1, + db_scale_4bit_12db_max), WSS_DOUBLE("Mic Playback Switch", 0, OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Mic Playback Volume", 0, - OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1), -WSS_DOUBLE("CD Playback Volume", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 1, 1, 15, 1), +WSS_DOUBLE_TLV("Mic Playback Volume", 0, + OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1, + db_scale_4bit_12db_max), +WSS_DOUBLE_TLV("CD Playback Volume", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 1, 1, 15, 1, + db_scale_4bit_12db_max), WSS_DOUBLE("Aux Playback Switch", 0, OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Aux Playback Volume", 0, - OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1), +WSS_DOUBLE_TLV("Aux Playback Volume", 0, + OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1, + db_scale_4bit_12db_max), }; static int __devinit snd_opti93x_mixer(struct snd_wss *chip) -- 1.6.4 ---------------------------------------------------------------------- Zbieraj punkty - wygrywaj nagrody! Kliknij >>> http://link.interia.pl/f24a8

2 1

[alsa-devel] dmix on hda-intel
by Norbert Preining 12 Dec '09

12 Dec '09

Dear all, (please cc) _From the wiki page: NOTE: For ALSA 1.0.9rc2 and higher you don't need to setup dmix. Dmix is enabled as default for soundcards which don't support hw mixing. My ALSA Version: 1.0.21 (see attached alsa-info output) Why is soft mixing still not working? Opening several audio clients does not work at all. Any suggestion, patches (-rc kernels, git diffs to apply?) Thanks Norbert ------------------------------------------------------------------------ Norbert Preining preining(a){jaist.ac.jp, logic.at, debian.org} JAIST, Japan TU Wien, Austria Debian TeX Task Force DSA: 0x09C5B094 fp: 14DF 2E6C 0307 BE6D AD76 A9C0 D2BF 4AA3 09C5 B094 ------------------------------------------------------------------------ You're one hundred percent positive that the ship which is crashed on the bottom of this ocean is the ship which you said you were one hundred percent positive could one hundred percent positively never crash? --- Douglas Adams, The Hitchhikers Guide to the Galaxy

2 1

[alsa-devel] [PATCH] dt019x: merge into the als100 driver
by Krzysztof Helt 11 Dec '09

11 Dec '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The als100 driver is so similar to the dt019x/als007 driver that one driver's source can be used for both drivers with only few changes. Merge the dt019x driver into the als100. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/Kconfig | 21 +--- sound/isa/Makefile | 2 - sound/isa/als100.c | 121 ++++++++++++++------ sound/isa/dt019x.c | 321 ---------------------------------------------------- 4 files changed, 90 insertions(+), 375 deletions(-) delete mode 100644 sound/isa/dt019x.c diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 02fe81c..8ebd070 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -63,15 +63,16 @@ config SND_AD1848 will be called snd-ad1848. config SND_ALS100 - tristate "Avance Logic ALS100/ALS120" + tristate "Diamond Tech. DT-019x and Avance Logic ALSxxx" depends on PNP select ISAPNP select SND_OPL3_LIB select SND_MPU401_UART select SND_SB16_DSP help - Say Y here to include support for soundcards based on Avance - Logic ALS100, ALS110, ALS120 and ALS200 chips. + Say Y here to include support for soundcards based on the + Diamond Technologies DT-019X or Avance Logic chips: ALS007, + ALS100, ALS110, ALS120 and ALS200 chips. To compile this driver as a module, choose M here: the module will be called snd-als100. @@ -127,20 +128,6 @@ config SND_CS4236 To compile this driver as a module, choose M here: the module will be called snd-cs4236. -config SND_DT019X - tristate "Diamond Technologies DT-019X, Avance Logic ALS-007" - depends on PNP - select ISAPNP - select SND_OPL3_LIB - select SND_MPU401_UART - select SND_SB16_DSP - help - Say Y here to include support for soundcards based on the - Diamond Technologies DT-019X or Avance Logic ALS-007 chips. - - To compile this driver as a module, choose M here: the module - will be called snd-dt019x. - config SND_ES968 tristate "Generic ESS ES968 driver" depends on PNP diff --git a/sound/isa/Makefile b/sound/isa/Makefile index b906b9a..c73d30c 100644 --- a/sound/isa/Makefile +++ b/sound/isa/Makefile @@ -7,7 +7,6 @@ snd-adlib-objs := adlib.o snd-als100-objs := als100.o snd-azt2320-objs := azt2320.o snd-cmi8330-objs := cmi8330.o -snd-dt019x-objs := dt019x.o snd-es18xx-objs := es18xx.o snd-opl3sa2-objs := opl3sa2.o snd-sc6000-objs := sc6000.o @@ -19,7 +18,6 @@ obj-$(CONFIG_SND_ADLIB) += snd-adlib.o obj-$(CONFIG_SND_ALS100) += snd-als100.o obj-$(CONFIG_SND_AZT2320) += snd-azt2320.o obj-$(CONFIG_SND_CMI8330) += snd-cmi8330.o -obj-$(CONFIG_SND_DT019X) += snd-dt019x.o obj-$(CONFIG_SND_ES18XX) += snd-es18xx.o obj-$(CONFIG_SND_OPL3SA2) += snd-opl3sa2.o obj-$(CONFIG_SND_SC6000) += snd-sc6000.o diff --git a/sound/isa/als100.c b/sound/isa/als100.c index 5fd52e4..20becc8 100644 --- a/sound/isa/als100.c +++ b/sound/isa/als100.c @@ -2,9 +2,13 @@ /* card-als100.c - driver for Avance Logic ALS100 based soundcards. Copyright (C) 1999-2000 by Massimo Piccioni <dafastidio(a)libero.it> + Copyright (C) 1999-2002 by Massimo Piccioni <dafastidio(a)libero.it> Thanks to Pierfrancesco 'qM2' Passerini. + Generalised for soundcards based on DT-0196 and ALS-007 chips + by Jonathan Woithe <jwoithe(a)physics.adelaide.edu.au>: June 2002. + 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 @@ -33,10 +37,10 @@ #define PFX "als100: " -MODULE_AUTHOR("Massimo Piccioni <dafastidio(a)libero.it>"); -MODULE_DESCRIPTION("Avance Logic ALS1X0"); -MODULE_LICENSE("GPL"); -MODULE_SUPPORTED_DEVICE("{{Avance Logic,ALS100 - PRO16PNP}," +MODULE_DESCRIPTION("Avance Logic ALS007/ALS1X0"); +MODULE_SUPPORTED_DEVICE("{{Diamond Technologies DT-019X}," + "{Avance Logic ALS-007}}" + "{{Avance Logic,ALS100 - PRO16PNP}," "{Avance Logic,ALS110}," "{Avance Logic,ALS120}," "{Avance Logic,ALS200}," @@ -45,9 +49,12 @@ MODULE_SUPPORTED_DEVICE("{{Avance Logic,ALS100 - PRO16PNP}," "{Avance Logic,ALS120}," "{RTL,RTL3000}}"); +MODULE_AUTHOR("Massimo Piccioni <dafastidio(a)libero.it>"); +MODULE_LICENSE("GPL"); + 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 int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ @@ -57,14 +64,15 @@ static int dma8[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* PnP setup */ static int dma16[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* PnP setup */ module_param_array(index, int, NULL, 0444); -MODULE_PARM_DESC(index, "Index value for als100 based soundcard."); +MODULE_PARM_DESC(index, "Index value for Avance Logic based soundcard."); module_param_array(id, charp, NULL, 0444); -MODULE_PARM_DESC(id, "ID string for als100 based soundcard."); +MODULE_PARM_DESC(id, "ID string for Avance Logic based soundcard."); module_param_array(enable, bool, NULL, 0444); -MODULE_PARM_DESC(enable, "Enable als100 based soundcard."); +MODULE_PARM_DESC(enable, "Enable Avance Logic based soundcard."); + +MODULE_ALIAS("snd-dt019x"); struct snd_card_als100 { - int dev_no; struct pnp_dev *dev; struct pnp_dev *devmpu; struct pnp_dev *devopl; @@ -72,25 +80,43 @@ struct snd_card_als100 { }; static struct pnp_card_device_id snd_als100_pnpids[] = { + /* DT197A30 */ + { .id = "RWB1688", + .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" } }, + .driver_data = SB_HW_DT019X }, + /* DT0196 / ALS-007 */ + { .id = "ALS0007", + .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" } }, + .driver_data = SB_HW_DT019X }, /* ALS100 - PRO16PNP */ - { .id = "ALS0001", .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" } } }, + { .id = "ALS0001", + .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" } }, + .driver_data = SB_HW_ALS100 }, /* ALS110 - MF1000 - Digimate 3D Sound */ - { .id = "ALS0110", .devs = { { "@@@1001" }, { "@X@1001" }, { "@H@1001" } } }, + { .id = "ALS0110", + .devs = { { "@@@1001" }, { "@X@1001" }, { "@H@1001" } }, + .driver_data = SB_HW_ALS100 }, /* ALS120 */ - { .id = "ALS0120", .devs = { { "@@@2001" }, { "@X@2001" }, { "@H@2001" } } }, + { .id = "ALS0120", + .devs = { { "@@@2001" }, { "@X@2001" }, { "@H@2001" } }, + .driver_data = SB_HW_ALS100 }, /* ALS200 */ - { .id = "ALS0200", .devs = { { "@@@0020" }, { "@X@0020" }, { "@H@0001" } } }, + { .id = "ALS0200", + .devs = { { "@@@0020" }, { "@X@0020" }, { "@H@0001" } }, + .driver_data = SB_HW_ALS100 }, /* ALS200 OEM */ - { .id = "ALS0200", .devs = { { "@@@0020" }, { "@X@0020" }, { "@H@0020" } } }, + { .id = "ALS0200", + .devs = { { "@@@0020" }, { "@X@0020" }, { "@H@0020" } }, + .driver_data = SB_HW_ALS100 }, /* RTL3000 */ - { .id = "RTL3000", .devs = { { "@@@2001" }, { "@X@2001" }, { "@H@2001" } } }, - { .id = "", } /* end */ + { .id = "RTL3000", + .devs = { { "@@@2001" }, { "@X@2001" }, { "@H@2001" } }, + .driver_data = SB_HW_ALS100 }, + { .id = "" } /* end */ }; MODULE_DEVICE_TABLE(pnp_card, snd_als100_pnpids); -#define DRIVER_NAME "snd-card-als100" - static int __devinit snd_card_als100_pnp(int dev, struct snd_card_als100 *acard, struct pnp_card_link *card, const struct pnp_card_device_id *id) @@ -113,8 +139,12 @@ static int __devinit snd_card_als100_pnp(int dev, struct snd_card_als100 *acard, return err; } port[dev] = pnp_port_start(pdev, 0); - dma8[dev] = pnp_dma(pdev, 1); - dma16[dev] = pnp_dma(pdev, 0); + if (id->driver_data == SB_HW_DT019X) + dma8[dev] = pnp_dma(pdev, 0); + else { + dma8[dev] = pnp_dma(pdev, 1); + dma16[dev] = pnp_dma(pdev, 0); + } irq[dev] = pnp_irq(pdev, 0); pdev = acard->devmpu; @@ -175,22 +205,33 @@ static int __devinit snd_card_als100_probe(int dev, } snd_card_set_dev(card, &pcard->card->dev); - if ((error = snd_sbdsp_create(card, port[dev], - irq[dev], - snd_sb16dsp_interrupt, - dma8[dev], - dma16[dev], - SB_HW_ALS100, &chip)) < 0) { + if (pid->driver_data == SB_HW_DT019X) + dma16[dev] = -1; + + error = snd_sbdsp_create(card, port[dev], irq[dev], + snd_sb16dsp_interrupt, + dma8[dev], dma16[dev], + pid->driver_data, + &chip); + if (error < 0) { snd_card_free(card); return error; } acard->chip = chip; - strcpy(card->driver, "ALS100"); - strcpy(card->shortname, "Avance Logic ALS100"); - sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d&%d", - card->shortname, chip->name, chip->port, - irq[dev], dma8[dev], dma16[dev]); + if (pid->driver_data == SB_HW_DT019X) { + strcpy(card->driver, "DT-019X"); + strcpy(card->shortname, "Diamond Tech. DT-019X"); + sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d", + card->shortname, chip->name, chip->port, + irq[dev], dma8[dev]); + } else { + strcpy(card->driver, "ALS100"); + strcpy(card->shortname, "Avance Logic ALS100"); + sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d&%d", + card->shortname, chip->name, chip->port, + irq[dev], dma8[dev], dma16[dev]); + } if ((error = snd_sb16dsp_pcm(chip, 0, NULL)) < 0) { snd_card_free(card); @@ -203,9 +244,19 @@ static int __devinit snd_card_als100_probe(int dev, } if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) { - if (snd_mpu401_uart_new(card, 0, MPU401_HW_ALS100, + int mpu_type = MPU401_HW_ALS100; + + if (mpu_irq[dev] == SNDRV_AUTO_IRQ) + mpu_irq[dev] = -1; + + if (pid->driver_data == SB_HW_DT019X) + mpu_type = MPU401_HW_MPU401; + + if (snd_mpu401_uart_new(card, 0, + mpu_type, mpu_port[dev], 0, - mpu_irq[dev], IRQF_DISABLED, + mpu_irq[dev], + mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0, NULL) < 0) snd_printk(KERN_ERR PFX "no MPU-401 device at 0x%lx\n", mpu_port[dev]); } @@ -291,7 +342,7 @@ static int snd_als100_pnp_resume(struct pnp_card_link *pcard) static struct pnp_card_driver als100_pnpc_driver = { .flags = PNP_DRIVER_RES_DISABLE, - .name = "als100", + .name = "als100", .id_table = snd_als100_pnpids, .probe = snd_als100_pnp_detect, .remove = __devexit_p(snd_als100_pnp_remove), @@ -312,7 +363,7 @@ static int __init alsa_card_als100_init(void) if (!als100_devices) { pnp_unregister_card_driver(&als100_pnpc_driver); #ifdef MODULE - snd_printk(KERN_ERR "no ALS100 based soundcards found\n"); + snd_printk(KERN_ERR "no Avance Logic based soundcards found\n"); #endif return -ENODEV; } diff --git a/sound/isa/dt019x.c b/sound/isa/dt019x.c deleted file mode 100644 index 80f5b1a..0000000 --- a/sound/isa/dt019x.c +++ /dev/null @@ -1,321 +0,0 @@ - -/* - dt019x.c - driver for Diamond Technologies DT-0197H based soundcards. - Copyright (C) 1999, 2002 by Massimo Piccioni <dafastidio(a)libero.it> - - Generalised for soundcards based on DT-0196 and ALS-007 chips - by Jonathan Woithe <jwoithe(a)physics.adelaide.edu.au>: June 2002. - - 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/wait.h> -#include <linux/pnp.h> -#include <linux/moduleparam.h> -#include <sound/core.h> -#include <sound/initval.h> -#include <sound/mpu401.h> -#include <sound/opl3.h> -#include <sound/sb.h> - -#define PFX "dt019x: " - -MODULE_AUTHOR("Massimo Piccioni <dafastidio(a)libero.it>"); -MODULE_DESCRIPTION("Diamond Technologies DT-019X / Avance Logic ALS-007"); -MODULE_LICENSE("GPL"); -MODULE_SUPPORTED_DEVICE("{{Diamond Technologies DT-019X}," - "{Avance Logic ALS-007}}"); - -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; /* Enable this card */ -static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ -static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ -static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ -static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* PnP setup */ -static int mpu_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 DT-019X based soundcard."); -module_param_array(id, charp, NULL, 0444); -MODULE_PARM_DESC(id, "ID string for DT-019X based soundcard."); -module_param_array(enable, bool, NULL, 0444); -MODULE_PARM_DESC(enable, "Enable DT-019X based soundcard."); - -struct snd_card_dt019x { - struct pnp_dev *dev; - struct pnp_dev *devmpu; - struct pnp_dev *devopl; - struct snd_sb *chip; -}; - -static struct pnp_card_device_id snd_dt019x_pnpids[] = { - /* DT197A30 */ - { .id = "RWB1688", .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" }, } }, - /* DT0196 / ALS-007 */ - { .id = "ALS0007", .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" }, } }, - { .id = "", } -}; - -MODULE_DEVICE_TABLE(pnp_card, snd_dt019x_pnpids); - - -#define DRIVER_NAME "snd-card-dt019x" - - -static int __devinit snd_card_dt019x_pnp(int dev, struct snd_card_dt019x *acard, - struct pnp_card_link *card, - const struct pnp_card_device_id *pid) -{ - struct pnp_dev *pdev; - int err; - - acard->dev = pnp_request_card_device(card, pid->devs[0].id, NULL); - if (acard->dev == NULL) - return -ENODEV; - - acard->devmpu = pnp_request_card_device(card, pid->devs[1].id, NULL); - acard->devopl = pnp_request_card_device(card, pid->devs[2].id, NULL); - - pdev = acard->dev; - - err = pnp_activate_dev(pdev); - if (err < 0) { - snd_printk(KERN_ERR PFX "DT-019X 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); - snd_printdd("dt019x: found audio interface: port=0x%lx, irq=0x%x, dma=0x%x\n", - port[dev],irq[dev],dma8[dev]); - - pdev = acard->devmpu; - if (pdev != NULL) { - err = pnp_activate_dev(pdev); - if (err < 0) { - pnp_release_card_device(pdev); - snd_printk(KERN_ERR PFX "DT-019X MPU401 pnp configure failure, skipping\n"); - goto __mpu_error; - } - mpu_port[dev] = pnp_port_start(pdev, 0); - mpu_irq[dev] = pnp_irq(pdev, 0); - snd_printdd("dt019x: found MPU-401: port=0x%lx, irq=0x%x\n", - mpu_port[dev],mpu_irq[dev]); - } else { - __mpu_error: - acard->devmpu = NULL; - mpu_port[dev] = -1; - } - - pdev = acard->devopl; - if (pdev != NULL) { - err = pnp_activate_dev(pdev); - if (err < 0) { - pnp_release_card_device(pdev); - snd_printk(KERN_ERR PFX "DT-019X OPL3 pnp configure failure, skipping\n"); - goto __fm_error; - } - fm_port[dev] = pnp_port_start(pdev, 0); - snd_printdd("dt019x: found OPL3 synth: port=0x%lx\n",fm_port[dev]); - } else { - __fm_error: - acard->devopl = NULL; - fm_port[dev] = -1; - } - - return 0; -} - -static int __devinit snd_card_dt019x_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_dt019x *acard; - struct snd_opl3 *opl3; - - error = snd_card_create(index[dev], id[dev], THIS_MODULE, - sizeof(struct snd_card_dt019x), &card); - if (error < 0) - return error; - acard = card->private_data; - - snd_card_set_dev(card, &pcard->card->dev); - if ((error = snd_card_dt019x_pnp(dev, acard, pcard, pid))) { - snd_card_free(card); - return error; - } - - if ((error = snd_sbdsp_create(card, port[dev], - irq[dev], - snd_sb16dsp_interrupt, - dma8[dev], - -1, - SB_HW_DT019X, - &chip)) < 0) { - snd_card_free(card); - return error; - } - acard->chip = chip; - - strcpy(card->driver, "DT-019X"); - strcpy(card->shortname, "Diamond Tech. DT-019X"); - sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d", - card->shortname, chip->name, chip->port, - irq[dev], dma8[dev]); - - if ((error = snd_sb16dsp_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 (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) { - if (mpu_irq[dev] == SNDRV_AUTO_IRQ) - mpu_irq[dev] = -1; - if (snd_mpu401_uart_new(card, 0, -/* MPU401_HW_SB,*/ - MPU401_HW_MPU401, - mpu_port[dev], 0, - mpu_irq[dev], - mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0, - NULL) < 0) - snd_printk(KERN_ERR PFX "no MPU-401 device at 0x%lx ?\n", mpu_port[dev]); - } - - if (fm_port[dev] > 0 && fm_port[dev] != SNDRV_AUTO_PORT) { - if (snd_opl3_create(card, - fm_port[dev], - fm_port[dev] + 2, - OPL3_HW_AUTO, 0, &opl3) < 0) { - snd_printk(KERN_ERR PFX "no OPL device at 0x%lx-0x%lx ?\n", - fm_port[dev], fm_port[dev] + 2); - } else { - if ((error = snd_opl3_timer_new(opl3, 0, 1)) < 0) { - snd_card_free(card); - return error; - } - if ((error = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) { - snd_card_free(card); - return error; - } - } - } - - 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 dt019x_devices; - -static int __devinit snd_dt019x_pnp_probe(struct pnp_card_link *card, - const struct pnp_card_device_id *pid) -{ - static int dev; - int res; - - for ( ; dev < SNDRV_CARDS; dev++) { - if (!enable[dev]) - continue; - res = snd_card_dt019x_probe(dev, card, pid); - if (res < 0) - return res; - dev++; - dt019x_devices++; - return 0; - } - return -ENODEV; -} - -static void __devexit snd_dt019x_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_dt019x_pnp_suspend(struct pnp_card_link *pcard, pm_message_t state) -{ - struct snd_card *card = pnp_get_card_drvdata(pcard); - struct snd_card_dt019x *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_dt019x_pnp_resume(struct pnp_card_link *pcard) -{ - struct snd_card *card = pnp_get_card_drvdata(pcard); - struct snd_card_dt019x *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 dt019x_pnpc_driver = { - .flags = PNP_DRIVER_RES_DISABLE, - .name = "dt019x", - .id_table = snd_dt019x_pnpids, - .probe = snd_dt019x_pnp_probe, - .remove = __devexit_p(snd_dt019x_pnp_remove), -#ifdef CONFIG_PM - .suspend = snd_dt019x_pnp_suspend, - .resume = snd_dt019x_pnp_resume, -#endif -}; - -static int __init alsa_card_dt019x_init(void) -{ - int err; - - err = pnp_register_card_driver(&dt019x_pnpc_driver); - if (err) - return err; - - if (!dt019x_devices) { - pnp_unregister_card_driver(&dt019x_pnpc_driver); -#ifdef MODULE - snd_printk(KERN_ERR "no DT-019X / ALS-007 based soundcards found\n"); -#endif - return -ENODEV; - } - return 0; -} - -static void __exit alsa_card_dt019x_exit(void) -{ - pnp_unregister_card_driver(&dt019x_pnpc_driver); -} - -module_init(alsa_card_dt019x_init) -module_exit(alsa_card_dt019x_exit) -- 1.6.4 ---------------------------------------------------------------------- Kup wlasne mieszkanie za 72 tys. zl. 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2 1

[alsa-devel] HD-Audio device unsuppored (Nvidia G2xx-series GPU implemented)
by Stefan Ringel 11 Dec '09

11 Dec '09

Hello Developers, I have an unsupported HD-audio device implemented in Nvidia G2xx-series GPU. (Vendor-ID 0x10de Device-ID 0x0be3, additional 0x0be2 and 0x0be4). I adding this id's in downloaded snapshot and tested it -> works. hda_intel.c ... { PCI_DEVICE(0x10de, 0x0d97), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0be2), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0be3), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0be4), .driver_data = AZX_DRIVER_NVIDIA }, /* Teradici */ ... ..., but the audio-codec 0x10de000b is also unsopported! In Nvidia Windows driver 1.00.00.59 from 2009.10.01 more alsa-unsupported audio-codecs id's. 4x /proc/asound/card1/codec#0 Codec: Nvidia ID b Address: 0 Function Id: 0x1 Vendor Id: 0x10de000b Subsystem Id: 0x10de0101 Revision Id: 0x100100 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x04 [Audio Output] wcaps 0x72b1: 8-Channels Digital Stripe CP Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Unsolicited: tag=00, enabled=0 Node 0x05 [Pin Complex] wcaps 0x407381: 8-Channels Digital CP Pincap 0x09000094: OUT Detect HDMI Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x04 Stefan Ringel wrote to alsa user mail list: I have a G210 Graphic-Card with implemented HD-Audio device. lspci list: 01:00.0 VGA compatible controller: nVidia Corporation GT200 [GeForce 210] (rev a2) 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 Interrupt: pin A routed to IRQ 16 Region 0: Memory at fb000000 (32-bit, non-prefetchable) [size=16M] Region 1: Memory at d0000000 (64-bit, prefetchable) [size=256M] Region 3: Memory at ee000000 (64-bit, prefetchable) [size=32M] Region 5: I/O ports at dc00 [size=128] [virtual] Expansion ROM at fcf00000 [disabled] [size=512K] Capabilities: [60] 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: [68] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [78] Express (v2) Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s unlimited, 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 512 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr- TransPend- LnkCap: Port #0, Speed 2.5GT/s, Width x16, ASPM L0s L1, Latency L0 <512ns, L1 <4us ClockPM+ Suprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; RCB 128 bytes Disabled- Retrain- CommClk+ ExtSynch- ClockPM+ AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x16, TrErr- Train- SlotClk+ DLActive- BWMgmt- ABWMgmt- Capabilities: [b4] Vendor Specific Information <?> Capabilities: [100] Virtual Channel <?> Capabilities: [128] Power Budgeting <?> Capabilities: [600] Vendor Specific Information <?> Kernel driver in use: nvidia Kernel modules: nvidia, nvidiafb 01:00.1 Audio device: nVidia Corporation Device 0be3 (rev a1) 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: 32 bytes Interrupt: pin A routed to IRQ 7 Region 0: Memory at fcffc000 (32-bit, non-prefetchable) [size=16K] Capabilities: [60] 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: [68] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [78] Express (v2) Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <4us, 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 512 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr- TransPend- LnkCap: Port #0, Speed 2.5GT/s, Width x16, ASPM L0s L1, Latency L0 <512ns, L1 <1us ClockPM+ Suprise- LLActRep- BwNot- LnkCtl: ASPM L0s L1 Enabled; RCB 128 bytes Disabled- Retrain- CommClk- ExtSynch- ClockPM+ AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x16, TrErr- Train- SlotClk+ DLActive- BWMgmt- ABWMgmt- Dark Shadow shadowofdarkness(a)gmail.com wrote to me: If there is any chance of this info helping here is my Laptop's unsupported hdmi audio device and the video card it is part of. Note the product id is a little different. 01:00.0 VGA compatible controller: nVidia Corporation GT215 [GeForce GTS 250M] (rev a2) Subsystem: Acer Incorporated [ALI] Device 026b 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 Interrupt: pin A routed to IRQ 16 Region 0: Memory at cc000000 (32-bit, non-prefetchable) [size=16M] Region 1: Memory at d0000000 (64-bit, prefetchable) [size=256M] Region 3: Memory at ce000000 (64-bit, prefetchable) [size=32M] Region 5: I/O ports at 2000 [size=128] [virtual] Expansion ROM at cd000000 [disabled] [size=512K] Capabilities: [60] 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: [68] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [78] Express (v2) Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s unlimited, 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 512 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr- TransPend- LnkCap: Port #0, Speed 2.5GT/s, Width x16, ASPM L0s L1, Latency L0 <256ns, L1 <4us ClockPM+ Suprise- LLActRep- BwNot- LnkCtl: ASPM L1 Enabled; RCB 128 bytes Disabled- Retrain- CommClk+ ExtSynch- ClockPM+ AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x1, TrErr- Train- SlotClk+ DLActive- BWMgmt- ABWMgmt- Capabilities: [b4] Vendor Specific Information <?> Capabilities: [100] Virtual Channel <?> Capabilities: [128] Power Budgeting <?> Capabilities: [600] Vendor Specific Information <?> Kernel driver in use: nvidia Kernel modules: nvidia, nvidiafb 01:00.1 Audio device: nVidia Corporation Device 0be4 (rev a1) Subsystem: Acer Incorporated [ALI] Device 026b 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 10 Region 0: Memory at cdefc000 (32-bit, non-prefetchable) [size=16K] Capabilities: [60] 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: [68] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [78] Express (v2) Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <4us, 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 512 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr- TransPend- LnkCap: Port #0, Speed 2.5GT/s, Width x16, ASPM L0s L1, Latency L0 <256ns, L1 <1us ClockPM+ Suprise- LLActRep- BwNot- LnkCtl: ASPM L0s L1 Enabled; RCB 128 bytes Disabled- Retrain- CommClk+ ExtSynch- ClockPM+ AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x1, TrErr- Train- SlotClk+ DLActive- BWMgmt- ABWMgmt-

1 0

[alsa-devel] [PATCH 0/5] Intel HDMI fixes
by Wu Fengguang 11 Dec '09

11 Dec '09

Hi Takashi, Here are some fixes for the Intel HDMI patch. I'm not perticular sure about the last two changes, review comments are highly appreciated. Thanks, Fengguang

2 10

[alsa-devel] ASoC update for 2.6.3
by Mark Brown 11 Dec '09

11 Dec '09

The following changes since commit 370066e2b13bafa8e742673f658e617b6ed143a4: Roel Kluin (1): ASoC: Wrong variable returned on error are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Olof Johansson (1): ASoC: Fix build of OMAP sound drivers sound/soc/omap/Makefile | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-)

2 1

[alsa-devel] Intel G45 8-Channel LPCM over HDMI
by Alexander Werner 11 Dec '09

11 Dec '09

Hi, I'm planning to build a linux-based, small sized PC for hearing aid acousticians. It would be highly unfavorable, if the connection between Receiver and PC was analog, because of the lower quality and the high number of cables. I searched for a appropriate mainboard, and found the DG45FC, which fits my needs perfectly, apart from the missing audio over HDMI support. Is there any plan to support multichannel LPCM output over HDMI on the Intel G45 platform? If not, is there anyone willing to implement HDMI support for the G45 chipset if I donate the needed hardware (permanently)? But if there's no chance to implement it in the near future, has anyone experience with an onboard chipset with multichannel LPCM over HDMI and ALSA support? Thanks, Alexander Werner

7 10

[alsa-devel] [PATCH] opti93x: move controls definitions to opti93x driver
by Krzysztof Helt 10 Dec '09

10 Dec '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Move OPTi93x controls definitions to the opti93x driver from the common wss-lib library module. These controls are used only by the opti93x driver. Also, fix capture source names. They are the same as opl3sa2 names. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/opti9xx/opti92x-ad1848.c | 85 ++++++++++++++++++++++++++++++++++++ sound/isa/wss/wss_lib.c | 80 ++++++--------------------------- 2 files changed, 100 insertions(+), 65 deletions(-) diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index d8eac3f..0991321 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -33,6 +33,7 @@ #include <asm/io.h> #include <asm/dma.h> #include <sound/core.h> +#include <sound/tlv.h> #include <sound/wss.h> #include <sound/mpu401.h> #include <sound/opl3.h> @@ -546,6 +547,85 @@ __skip_mpu: #ifdef OPTi93X +static const DECLARE_TLV_DB_SCALE(db_scale_6bit, -9450, 150, 0); + +static struct snd_kcontrol_new snd_opti93x_controls[] = { +WSS_DOUBLE("Master Playback Switch", 0, + OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1), +WSS_DOUBLE_TLV("Master Playback Volume", 0, + OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1, + db_scale_6bit), +WSS_DOUBLE("PCM Playback Volume", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 31, 1), +WSS_DOUBLE("FM Playback Volume", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 1, 1, 15, 1), +WSS_DOUBLE("Line Playback Switch", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), +WSS_DOUBLE("Line Playback Volume", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 15, 1), +WSS_DOUBLE("Mic Playback Switch", 0, + OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 7, 7, 1, 1), +WSS_DOUBLE("Mic Playback Volume", 0, + OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1), +WSS_DOUBLE("CD Playback Volume", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 1, 1, 15, 1), +WSS_DOUBLE("Aux Playback Switch", 0, + OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 7, 7, 1, 1), +WSS_DOUBLE("Aux Playback Volume", 0, + OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1), +}; + +static int __devinit snd_opti93x_mixer(struct snd_wss *chip) +{ + struct snd_card *card; + unsigned int idx; + struct snd_ctl_elem_id id1, id2; + int err; + + if (snd_BUG_ON(!chip || !chip->pcm)) + return -EINVAL; + + card = chip->card; + + strcpy(card->mixername, chip->pcm->name); + + memset(&id1, 0, sizeof(id1)); + memset(&id2, 0, sizeof(id2)); + id1.iface = id2.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + /* reassign AUX0 switch to CD */ + strcpy(id1.name, "Aux Playback Switch"); + strcpy(id2.name, "CD Playback Switch"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) { + snd_printk(KERN_ERR "Cannot rename opti93x control\n"); + return err; + } + /* reassign AUX1 switch to FM */ + strcpy(id1.name, "Aux Playback Switch"); id1.index = 1; + strcpy(id2.name, "FM Playback Switch"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) { + snd_printk(KERN_ERR "Cannot rename opti93x control\n"); + return err; + } + /* remove AUX1 volume */ + strcpy(id1.name, "Aux Playback Volume"); id1.index = 1; + snd_ctl_remove_id(card, &id1); + + /* Replace WSS volume controls with OPTi93x volume controls */ + id1.index = 0; + for (idx = 0; idx < ARRAY_SIZE(snd_opti93x_controls); idx++) { + strcpy(id1.name, snd_opti93x_controls[idx].name); + snd_ctl_remove_id(card, &id1); + + err = snd_ctl_add(card, + snd_ctl_new1(&snd_opti93x_controls[idx], chip)); + if (err < 0) + return err; + } + return 0; +} + static irqreturn_t snd_opti93x_interrupt(int irq, void *dev_id) { struct snd_opti9xx *chip = dev_id; @@ -754,6 +834,11 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) error = snd_wss_mixer(codec); if (error < 0) return error; +#ifdef OPTi93X + error = snd_opti93x_mixer(codec); + if (error < 0) + return error; +#endif #ifdef CS4231 error = snd_wss_timer(codec, 0, &timer); if (error < 0) diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 5b9d6c1..9191b32 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -2014,6 +2014,7 @@ static int snd_wss_info_mux(struct snd_kcontrol *kcontrol, case WSS_HW_INTERWAVE: ptexts = gusmax_texts; break; + case WSS_HW_OPTI93X: case WSS_HW_OPL3SA2: ptexts = opl3sa_texts; break; @@ -2246,54 +2247,12 @@ WSS_SINGLE("Beep Bypass Playback Switch", 0, CS4231_MONO_CTRL, 5, 1, 0), }; -static struct snd_kcontrol_new snd_opti93x_controls[] = { -WSS_DOUBLE("Master Playback Switch", 0, - OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1), -WSS_DOUBLE_TLV("Master Playback Volume", 0, - OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1, - db_scale_6bit), -WSS_DOUBLE("PCM Playback Switch", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), -WSS_DOUBLE("PCM Playback Volume", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 31, 1), -WSS_DOUBLE("FM Playback Switch", 0, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("FM Playback Volume", 0, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 1, 1, 15, 1), -WSS_DOUBLE("Line Playback Switch", 0, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), -WSS_DOUBLE("Line Playback Volume", 0, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 15, 1), -WSS_DOUBLE("Mic Playback Switch", 0, - OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Mic Playback Volume", 0, - OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1), -WSS_DOUBLE("Mic Boost", 0, - CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), -WSS_DOUBLE("CD Playback Switch", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("CD Playback Volume", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 1, 1, 15, 1), -WSS_DOUBLE("Aux Playback Switch", 0, - OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Aux Playback Volume", 0, - OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1), -WSS_DOUBLE("Capture Volume", 0, - CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), -{ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = snd_wss_info_mux, - .get = snd_wss_get_mux, - .put = snd_wss_put_mux, -} -}; - int snd_wss_mixer(struct snd_wss *chip) { struct snd_card *card; unsigned int idx; int err; + int count = ARRAY_SIZE(snd_wss_controls); if (snd_BUG_ON(!chip || !chip->pcm)) return -EINVAL; @@ -2302,28 +2261,19 @@ int snd_wss_mixer(struct snd_wss *chip) strcpy(card->mixername, chip->pcm->name); - if (chip->hardware == WSS_HW_OPTI93X) - for (idx = 0; idx < ARRAY_SIZE(snd_opti93x_controls); idx++) { - err = snd_ctl_add(card, - snd_ctl_new1(&snd_opti93x_controls[idx], - chip)); - if (err < 0) - return err; - } - else { - int count = ARRAY_SIZE(snd_wss_controls); - - /* Use only the first 11 entries on AD1848 */ - if (chip->hardware & WSS_HW_AD1848_MASK) - count = 11; - - for (idx = 0; idx < count; idx++) { - err = snd_ctl_add(card, - snd_ctl_new1(&snd_wss_controls[idx], - chip)); - if (err < 0) - return err; - } + /* Use only the first 11 entries on AD1848 */ + if (chip->hardware & WSS_HW_AD1848_MASK) + count = 11; + /* There is no loopback on OPTI93X */ + else if (chip->hardware == WSS_HW_OPTI93X) + count = 9; + + for (idx = 0; idx < count; idx++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_wss_controls[idx], + chip)); + if (err < 0) + return err; } return 0; } -- 1.6.4 ---------------------------------------------------------------------- Internetowi nie placa! Otworz Konto Direct. http://link.interia.pl/f24f7

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[alsa-devel] [PATCH] Fix build of OMAP sound drivers
by Olof Johansson 10 Dec '09

10 Dec '09

There are build errors when building for some of the omap2/3 boards without enabling sound: sound/built-in.o:(.data+0x43bc): undefined reference to `soc_codec_dev_tlv320aic23' sound/built-in.o:(.data+0x43cc): undefined reference to `tlv320aic23_dai' Confused me quite a bit since the drivers that had references to the codec weren't enabled. Turns out the Makefile was using the wrong config option to enable them. Patch below. Reported-by: Anand Gadiyar <gadiyar(a)ti.com> Signed-off-by: Olof Johansson <olof(a)lixom.net> diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index d49458a..3db8a6c 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -23,9 +23,9 @@ obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(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 -obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o -obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o -obj-$(CONFIG_MACH_OMAP3517EVM) += snd-soc-am3517evm.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP3517EVM) += snd-soc-am3517evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o

3 2

[alsa-devel] HD-Audio device unsuppored (Nvidia G2xx-series GPU implemented)
by Stefan Ringel 10 Dec '09

10 Dec '09

Hello Developers, I have an unsupported HD-audio device implemented in Nvidia G2xx-series GPU. (Vendor-ID 0x10de Device-ID 0x0be3, additional 0x0be2 and 0x0be4). I adding this id's in downloaded snapshot and tested it -> works. hda_intel.c ... { PCI_DEVICE(0x10de, 0x0d97), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0be2), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0be3), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0be4), .driver_data = AZX_DRIVER_NVIDIA }, /* Teradici */ ... ..., but the audio-codec 0x10de000b is also unsopported! In Nvidia Windows driver 1.00.00.59 from 2009.10.01 more alsa-unsupported audio-codecs id's. 4x /proc/asound/card1/codec#0 Codec: Nvidia ID b Address: 0 Function Id: 0x1 Vendor Id: 0x10de000b Subsystem Id: 0x10de0101 Revision Id: 0x100100 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x04 [Audio Output] wcaps 0x72b1: 8-Channels Digital Stripe CP Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Unsolicited: tag=00, enabled=0 Node 0x05 [Pin Complex] wcaps 0x407381: 8-Channels Digital CP Pincap 0x09000094: OUT Detect HDMI Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x04 Stefan Ringel wrote to alsa user mail list: I have a G210 Graphic-Card with implemented HD-Audio device. lspci list: 01:00.0 VGA compatible controller: nVidia Corporation GT200 [GeForce 210] (rev a2) 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 Interrupt: pin A routed to IRQ 16 Region 0: Memory at fb000000 (32-bit, non-prefetchable) [size=16M] Region 1: Memory at d0000000 (64-bit, prefetchable) [size=256M] Region 3: Memory at ee000000 (64-bit, prefetchable) [size=32M] Region 5: I/O ports at dc00 [size=128] [virtual] Expansion ROM at fcf00000 [disabled] [size=512K] Capabilities: [60] 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: [68] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [78] Express (v2) Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s unlimited, 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 512 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr- TransPend- LnkCap: Port #0, Speed 2.5GT/s, Width x16, ASPM L0s L1, Latency L0 <512ns, L1 <4us ClockPM+ Suprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; RCB 128 bytes Disabled- Retrain- CommClk+ ExtSynch- ClockPM+ AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x16, TrErr- Train- SlotClk+ DLActive- BWMgmt- ABWMgmt- Capabilities: [b4] Vendor Specific Information <?> Capabilities: [100] Virtual Channel <?> Capabilities: [128] Power Budgeting <?> Capabilities: [600] Vendor Specific Information <?> Kernel driver in use: nvidia Kernel modules: nvidia, nvidiafb 01:00.1 Audio device: nVidia Corporation Device 0be3 (rev a1) 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: 32 bytes Interrupt: pin A routed to IRQ 7 Region 0: Memory at fcffc000 (32-bit, non-prefetchable) [size=16K] Capabilities: [60] 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: [68] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [78] Express (v2) Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <4us, 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 512 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr- TransPend- LnkCap: Port #0, Speed 2.5GT/s, Width x16, ASPM L0s L1, Latency L0 <512ns, L1 <1us ClockPM+ Suprise- LLActRep- BwNot- LnkCtl: ASPM L0s L1 Enabled; RCB 128 bytes Disabled- Retrain- CommClk- ExtSynch- ClockPM+ AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x16, TrErr- Train- SlotClk+ DLActive- BWMgmt- ABWMgmt- Dark Shadow shadowofdarkness(a)gmail.com wrote to me: If there is any chance of this info helping here is my Laptop's unsupported hdmi audio device and the video card it is part of. Note the product id is a little different. 01:00.0 VGA compatible controller: nVidia Corporation GT215 [GeForce GTS 250M] (rev a2) Subsystem: Acer Incorporated [ALI] Device 026b 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 Interrupt: pin A routed to IRQ 16 Region 0: Memory at cc000000 (32-bit, non-prefetchable) [size=16M] Region 1: Memory at d0000000 (64-bit, prefetchable) [size=256M] Region 3: Memory at ce000000 (64-bit, prefetchable) [size=32M] Region 5: I/O ports at 2000 [size=128] [virtual] Expansion ROM at cd000000 [disabled] [size=512K] Capabilities: [60] 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: [68] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [78] Express (v2) Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s unlimited, 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 512 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr- TransPend- LnkCap: Port #0, Speed 2.5GT/s, Width x16, ASPM L0s L1, Latency L0 <256ns, L1 <4us ClockPM+ Suprise- LLActRep- BwNot- LnkCtl: ASPM L1 Enabled; RCB 128 bytes Disabled- Retrain- CommClk+ ExtSynch- ClockPM+ AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x1, TrErr- Train- SlotClk+ DLActive- BWMgmt- ABWMgmt- Capabilities: [b4] Vendor Specific Information <?> Capabilities: [100] Virtual Channel <?> Capabilities: [128] Power Budgeting <?> Capabilities: [600] Vendor Specific Information <?> Kernel driver in use: nvidia Kernel modules: nvidia, nvidiafb 01:00.1 Audio device: nVidia Corporation Device 0be4 (rev a1) Subsystem: Acer Incorporated [ALI] Device 026b 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 10 Region 0: Memory at cdefc000 (32-bit, non-prefetchable) [size=16K] Capabilities: [60] 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: [68] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [78] Express (v2) Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <4us, 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 512 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr- TransPend- LnkCap: Port #0, Speed 2.5GT/s, Width x16, ASPM L0s L1, Latency L0 <256ns, L1 <1us ClockPM+ Suprise- LLActRep- BwNot- LnkCtl: ASPM L0s L1 Enabled; RCB 128 bytes Disabled- Retrain- CommClk+ ExtSynch- ClockPM+ AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x1, TrErr- Train- SlotClk+ DLActive- BWMgmt- ABWMgmt-

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[alsa-devel] [PATCH 1/2] ASoC: Add DA7210 codec device support for ALSA
by Kuninori Morimoto 10 Dec '09

10 Dec '09

This original driver was created by Dialog Semiconductor, and cleanuped by Kuninori Morimoto. Special thanks to David Chen. This is very simple ASoC codec driver. It was tested by EcoVec24 board. Signed-off-by: David Chen <Dajun.chen(a)diasemi.com> Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/da7210.c | 773 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/da7210.h | 24 ++ 4 files changed, 803 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/da7210.c create mode 100644 sound/soc/codecs/da7210.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 52b005f..ff33b02 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -23,6 +23,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4671 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_MAX9877 if I2C + select SND_SOC_DA7210 if I2C select SND_SOC_PCM3008 select SND_SOC_SPDIF select SND_SOC_SSM2602 if I2C @@ -112,6 +113,9 @@ config SND_SOC_AK4671 config SND_SOC_CS4270 tristate +config SND_SOC_DA7210 + tristate + # Cirrus Logic CS4270 Codec VD = 3.3V Errata # Select if you are affected by the errata where the part will not function # if MCLK divide-by-1.5 is selected and VD is set to 3.3V. The driver will diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index dbaecb1..a495bf8 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -10,6 +10,7 @@ snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o snd-soc-cs4270-objs := cs4270.o snd-soc-cx20442-objs := cx20442.o +snd-soc-da7210-objs := da7210.o snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-spdif-objs := spdif_transciever.o @@ -66,6 +67,7 @@ obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o +obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c new file mode 100644 index 0000000..1356b90 --- /dev/null +++ b/sound/soc/codecs/da7210.c @@ -0,0 +1,773 @@ +/* + * DA7210 ALSA Soc codec driver + * + * Copyright (c) 2009 Dialog Semiconductor + * Written by David Chen <Dajun.chen(a)diasemi.com> + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Cleanups by Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * Tested on SuperH Ecovec24 board with S16/S24 LE in 48KHz using I2S + * + * 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/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/initval.h> +#include <asm/div64.h> + +#include "da7210.h" + +/* DA7210 register space */ +#define DA7210_STATUS 0x02 +#define DA7210_STARTUP1 0x03 +#define DA7210_STARTUP2 0x04 +#define DA7210_STARTUP3 0x05 +#define DA7210_MIC_L 0x07 +#define DA7210_MIC_R 0x08 +#define DA7210_IN_GAIN 0x0C +#define DA7210_INMIX_L 0x0D +#define DA7210_INMIX_R 0x0E +#define DA7210_ADC_HPF 0x0F +#define DA7210_ADC 0x10 +#define DA7210_DAC_HPF 0x14 +#define DA7210_DAC_L 0x15 +#define DA7210_DAC_R 0x16 +#define DA7210_DAC_SEL 0x17 +#define DA7210_OUTMIX_L 0x1C +#define DA7210_OUTMIX_R 0x1D +#define DA7210_OUT2 0x20 +#define DA7210_HP_L_VOL 0x21 +#define DA7210_HP_R_VOL 0x22 +#define DA7210_HP_CFG 0x23 +#define DA7210_DAI_SRC_SEL 0x25 +#define DA7210_DAI_CFG1 0x26 +#define DA7210_DAI_CFG3 0x28 +#define DA7210_PLL_DIV3 0x2B +#define DA7210_PLL 0x2C + +/* STARTUP1 bit fields */ +#define DA7210_SC_MST_EN (1 << 0) + +/* STARTUP2 bit fields */ +#define DA7210_LOUT1_L_STBY (1 << 0) +#define DA7210_LOUT1_R_STBY (1 << 1) +#define DA7210_LOUT2_STBY (1 << 2) +#define DA7210_HP_L_STBY (1 << 3) +#define DA7210_HP_R_STBY (1 << 4) +#define DA7210_DAC_L_STBY (1 << 5) +#define DA7210_DAC_R_STBY (1 << 6) + +/* STARTUP3 bit fields */ +#define DA7210_MIC_L_STBY (1 << 0) +#define DA7210_MIC_R_STBY (1 << 1) +#define DA7210_LIN1_L_STBY (1 << 2) +#define DA7210_LIN1_R_STBY (1 << 3) +#define DA7210_LIN2_STBY (1 << 4) +#define DA7210_ADC_L_STBY (1 << 5) +#define DA7210_ADC_R_STBY (1 << 6) + +/* MIC_L bit fields */ +#define DA7210_MICBIAS_EN (1 << 6) +#define DA7210_MIC_L_EN (1 << 7) + +/* MIC_R bit fields */ +#define DA7210_MIC_R_EN (1 << 7) + +/* INMIX_L bit fields */ +#define DA7210_IN_L_EN (1 << 7) + +/* INMIX_R bit fields */ +#define DA7210_IN_R_EN (1 << 7) + +/* ADC_HPF bit fields */ +#define DA7210_ADC_HPF_EN (1 << 3) +#define DA7210_ADC_VOICE_EN (1 << 7) + +/* ADC bit fields */ +#define DA7210_ADC_L_EN (1 << 3) +#define DA7210_ADC_R_EN (1 << 7) + +/* DAC_HPF bit fields */ +#define DA7210_DAC_HPF_EN (1 << 3) +#define DA7210_DAC_VOICE_EN (1 << 7) + +/* DAC_SEL bit fields */ +#define DA7210_DAC_L_SRC_DAI_L (4 << 0) +#define DA7210_DAC_L_EN (1 << 3) +#define DA7210_DAC_R_SRC_DAI_R (5 << 4) +#define DA7210_DAC_R_EN (1 << 7) + +/* OUTMIX_L bit fields */ +#define DA7210_OUT_L_EN (1 << 7) + +/* OUTMIX_R bit fields */ +#define DA7210_OUT_R_EN (1 << 7) + +/* HP_CFG bit fields */ +#define DA7210_HP_2CAP_MODE (1 << 1) +#define DA7210_HP_SENSE_EN (1 << 2) +#define DA7210_HP_L_EN (1 << 3) +#define DA7210_HP_MODE (1 << 6) +#define DA7210_HP_R_EN (1 << 7) + +/* DAI_SRC_SEL bit fields */ +#define DA7210_DAI_OUT_L_SRC (6 << 0) +#define DA7210_DAI_OUT_R_SRC (7 << 4) + +/* DAI_CFG1 bit fields */ +#define DA7210_DAI_WORD_S16_LE (0 << 0) +#define DA7210_DAI_WORD_S24_LE (2 << 0) +#define DA7210_DAI_FLEN_64BIT (1 << 2) +#define DA7210_DAI_MODE_MASTER (1 << 7) + +/* DAI_CFG3 bit fields */ +#define DA7210_DAI_FORMAT_I2SMODE (0 << 0) +#define DA7210_DAI_OE (1 << 3) +#define DA7210_DAI_EN (1 << 7) + +/*PLL_DIV3 bit fields */ +#define DA7210_MCLK_RANGE_10_20_MHZ (1 << 4) +#define DA7210_PLL_BYP (1 << 6) + +/* PLL bit fields */ +#define DA7210_PLL_FS_48000 (11 << 0) + +#define DA7210_VERSION "0.0.1" + +/* Codec private data */ +struct da7210_priv { + struct snd_soc_codec codec; +}; + +static struct snd_soc_codec *da7210_codec; + +/* + * Register cache + */ +static const u8 da7210_reg[] = { + 0x00, 0x11, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R0 - R7 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x08, /* R8 - RF */ + 0x00, 0x00, 0x00, 0x00, 0x08, 0x10, 0x10, 0x54, /* R10 - R17 */ + 0x40, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R18 - R1F */ + 0x00, 0x00, 0x00, 0x02, 0x00, 0x76, 0x00, 0x00, /* R20 - R27 */ + 0x04, 0x00, 0x00, 0x30, 0x2A, 0x00, 0x40, 0x00, /* R28 - R2F */ + 0x40, 0x00, 0x40, 0x00, 0x40, 0x00, 0x40, 0x00, /* R30 - R37 */ + 0x40, 0x00, 0x40, 0x00, 0x40, 0x00, 0x00, 0x00, /* R38 - R3F */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R40 - R4F */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R48 - R4F */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R50 - R57 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R58 - R5F */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R60 - R67 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R68 - R6F */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R70 - R77 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x54, 0x54, 0x00, /* R78 - R7F */ + 0x00, 0x00, 0x2C, 0x00, 0x00, 0x00, 0x00, 0x00, /* R80 - R87 */ + 0x00, /* R88 */ +}; + +/* + * Read da7210 register cache + */ +static inline unsigned int da7210_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + BUG_ON(reg > ARRAY_SIZE(da7210_reg)); + return cache[reg]; +} + +/* + * Write da7210 register cache + */ +static inline void da7210_write_reg_cache(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + u8 *cache = codec->reg_cache; + + cache[reg] = value; +} + +/* + * Write to the da7210 register space + */ +static int da7210_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + + u8 data[2]; + + /* data[0] da7210 register offset */ + /* data[1] register data */ + data[0] = reg & 0xff; + data[1] = value & 0xff; + + if (2 != codec->hw_write(codec->control_data, data, 2)) { + dev_warn(codec->dev, "I2C write Failed!\n"); + return -EIO; + } + + /* write the cache only if hardware write is successful */ + if (data[0] < ARRAY_SIZE(da7210_reg)) + da7210_write_reg_cache(codec, data[0], data[1]); + + return 0; +} + +/* + * Read from the da7210 register space. + */ +static inline unsigned int da7210_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + /* FIXME !! + * + * we should read status from I2C. + * But not supported now. + */ + if (DA7210_STATUS == reg) + return -EIO; + + return da7210_read_reg_cache(codec, reg); +} + +static const char *da7210_mic_bias_voltage[] = { "1.5", "1.6", "2.2", "2.3" }; + +static const struct soc_enum da7210_enum[] = { + SOC_ENUM_SINGLE(DA7210_MIC_L, 4, 4, da7210_mic_bias_voltage), +}; + +/* Add non DAPM controls */ +static const struct snd_kcontrol_new da7210_snd_controls[] = { + /* Mixer Playback controls */ + SOC_DOUBLE_R("DAC Gain", DA7210_DAC_L, DA7210_DAC_R, 0, 0x37, 1), + SOC_DOUBLE_R("HeadPhone Playback Volume", + DA7210_HP_L_VOL, DA7210_HP_R_VOL, 0, 0x3f, 0), + /* Mixer Capture controls */ + SOC_DOUBLE_R("Mic Capture Volume", + DA7210_MIC_L, DA7210_MIC_R, 0, 0x07, 0), + SOC_DOUBLE("In PGA Gain", DA7210_IN_GAIN, 0, 4, 0x0F, 0), + SOC_SINGLE("Mic Bias", DA7210_MIC_L, 6, 1, 0), + SOC_ENUM("Mic Bias Voltage", da7210_enum[0]), +}; + +/* ----------------------------Capture Mixers--------------------------- */ + +/* In Mixer Left */ +static const struct snd_kcontrol_new da7210_in_left_mixer_controls[] = { + SOC_DAPM_SINGLE("MIC_L Switch", DA7210_INMIX_L, 0, 1, 0), + SOC_DAPM_SINGLE("MIC_R Switch", DA7210_INMIX_L, 1, 1, 0), + SOC_DAPM_SINGLE("Aux1_L Switch", DA7210_INMIX_L, 2, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_INMIX_L, 3, 1, 0), + SOC_DAPM_SINGLE("DAC_L Switch", DA7210_INMIX_L, 4, 1, 0), +}; + +/* In Mixer Right */ +static const struct snd_kcontrol_new da7210_in_right_mixer_controls[] = { + SOC_DAPM_SINGLE("MIC_R Switch", DA7210_INMIX_R, 0, 1, 0), + SOC_DAPM_SINGLE("MIC_L Switch", DA7210_INMIX_R, 1, 1, 0), + SOC_DAPM_SINGLE("Aux1_R Switch", DA7210_INMIX_R, 2, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_INMIX_R, 3, 1, 0), + SOC_DAPM_SINGLE("DAC_R Switch", DA7210_INMIX_R, 4, 1, 0), + SOC_DAPM_SINGLE("INPGA_L Switch", DA7210_INMIX_R, 5, 1, 0), +}; + +/*----------------------------Playback Mixers---------------------------*/ +/* Out Mixer Left */ +static const struct snd_kcontrol_new da7210_out_mixer_left_controls[] = { + SOC_DAPM_SINGLE("AUX1_L Switch", DA7210_OUTMIX_L, 0, 1, 0), + SOC_DAPM_SINGLE("AUX2 Switch", DA7210_OUTMIX_L, 1, 1, 0), + SOC_DAPM_SINGLE("IN_L Switch", DA7210_OUTMIX_L, 2, 1, 0), + SOC_DAPM_SINGLE("IN_R Switch", DA7210_OUTMIX_L, 3, 1, 0), + SOC_DAPM_SINGLE("DAC_L Switch", DA7210_OUTMIX_L, 4, 1, 0), +}; + +/* Out Mixer Right */ +static const struct snd_kcontrol_new da7210_out_mixer_right_controls[] = { + SOC_DAPM_SINGLE("AUX1_R Switch", DA7210_OUTMIX_R, 0, 1, 0), + SOC_DAPM_SINGLE("AUX2 Switch", DA7210_OUTMIX_R, 1, 1, 0), + SOC_DAPM_SINGLE("IN_L Switch", DA7210_OUTMIX_R, 2, 1, 0), + SOC_DAPM_SINGLE("IN_R Switch", DA7210_OUTMIX_R, 3, 1, 0), + SOC_DAPM_SINGLE("DAC_R Switch", DA7210_OUTMIX_R, 4, 1, 0), +}; + +/* Mono Mixer */ +static const struct snd_kcontrol_new da7210_mono_mixer_controls[] = { + SOC_DAPM_SINGLE("IN_L Switch", DA7210_OUT2, 3, 1, 0), + SOC_DAPM_SINGLE("IN_R Switch", DA7210_OUT2, 4, 1, 0), + SOC_DAPM_SINGLE("DAC_L Switch", DA7210_OUT2, 5, 1, 0), + SOC_DAPM_SINGLE("DAC_R Switch", DA7210_OUT2, 6, 1, 0), +}; + +static const struct snd_kcontrol_new da7210_headphone_left_control = + SOC_DAPM_SINGLE("Switch", DA7210_STARTUP2, 3, 1, 1); +static const struct snd_kcontrol_new da7210_headphone_right_control = + SOC_DAPM_SINGLE("Switch", DA7210_STARTUP2, 4, 1, 1); +static const struct snd_kcontrol_new da7210_MicIn_left_control = + SOC_DAPM_SINGLE("Switch", DA7210_STARTUP3, 0, 1, 1); +static const struct snd_kcontrol_new da7210_MicIn_right_control = + SOC_DAPM_SINGLE("Switch", DA7210_STARTUP3, 1, 1, 1); + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget da7210_dapm_widgets[] = { + /* DAMP stream domain - DAC. Enabled when playback is started */ + SND_SOC_DAPM_DAC("DAC Left", "Playback", DA7210_STARTUP2, 5, 1), + SND_SOC_DAPM_DAC("DAC Right", "Playback", DA7210_STARTUP2, 6, 1), + + /* DAMP stream domain - ADC. Enabled when capture is started */ + SND_SOC_DAPM_ADC("ADC Left", "Capture", DA7210_STARTUP3, 5, 1), + SND_SOC_DAPM_ADC("ADC Right", "Capture", DA7210_STARTUP3, 6, 1), + + /* DAPM path domain - switches and mixers */ + /* Automatically set when mixer settings are changed by the user */ + SND_SOC_DAPM_SWITCH("HPL Enable", SND_SOC_NOPM, 0, 0, + &da7210_headphone_left_control), + SND_SOC_DAPM_SWITCH("HPR Enable", SND_SOC_NOPM, 0, 0, + &da7210_headphone_right_control), + SND_SOC_DAPM_SWITCH("MicL Enable", SND_SOC_NOPM, 0, 0, + &da7210_MicIn_left_control), + SND_SOC_DAPM_SWITCH("MicR Enable", SND_SOC_NOPM, 0, 0, + &da7210_MicIn_right_control), + + SND_SOC_DAPM_MIXER("Out Mixer Left", SND_SOC_NOPM, 0, 0, + &da7210_out_mixer_left_controls[0], + ARRAY_SIZE(da7210_out_mixer_left_controls)), + + SND_SOC_DAPM_MIXER("Out Mixer Right", SND_SOC_NOPM, 0, 0, + &da7210_out_mixer_right_controls[0], + ARRAY_SIZE(da7210_out_mixer_right_controls)), + + SND_SOC_DAPM_MIXER("In Mixer Left", SND_SOC_NOPM, 0, 0, + &da7210_in_left_mixer_controls[0], + ARRAY_SIZE(da7210_in_left_mixer_controls)), + + SND_SOC_DAPM_MIXER("In Mixer Right", SND_SOC_NOPM, 0, 0, + &da7210_in_right_mixer_controls[0], + ARRAY_SIZE(da7210_in_right_mixer_controls)), + + /* DAPM Platform domain. Physically connected input and ouput pins */ + SND_SOC_DAPM_OUTPUT("HPL"), /*Headphone Out left*/ + SND_SOC_DAPM_OUTPUT("HPR"), /*Headphone out Right*/ + SND_SOC_DAPM_INPUT("MICL"), /*MicIn left*/ + SND_SOC_DAPM_INPUT("MICR"), /*MicIn Right*/ +}; + +/* DAPM audio route definition */ +static const struct snd_soc_dapm_route audio_map[] = { + /*Out Mixer Left*/ + {"Out Mixer Left", "DAC_L Switch", "DAC Left"}, + + /*Out Mixer Right*/ + {"Out Mixer Right", "DAC_R Switch", "DAC Right"}, + + /*In Mixer Left*/ + {"In Mixer Left", "MIC_L Switch", "MicL Enable"}, + + /*In Mixer Right*/ + {"In Mixer Right", "MIC_R Switch", "MicR Enable"}, + + /*HPL*/ + {"HPL", NULL, "HPL Enable"}, + {"HPL Enable", "Switch", "Out Mixer Left"}, + + /*HPR*/ + {"HPR", NULL, "HPR Enable"}, + {"HPR Enable", "Switch", "Out Mixer Right"}, + + /*MICL*/ + {"ADC Left", NULL, "In Mixer Left"}, + {"MicL Enable", "Switch", "MICL"}, + + /*MICR*/ + {"ADC Right", NULL, "In Mixer Right"}, + {"MicR Enable", "Switch", "MICR"}, +}; + +static int da7210_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, da7210_dapm_widgets, + ARRAY_SIZE(da7210_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_new_widgets(codec); + return 0; +} + +/* + * Set PCM DAI word length. + * Enable Voice Filter if Fs <= 16KHz + */ +static int da7210_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + unsigned int dai_cfg1; + unsigned int value, reg, mask; + + dai_cfg1 = 0xFC & da7210_read(codec, DA7210_DAI_CFG1); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + dai_cfg1 |= DA7210_DAI_WORD_S16_LE; + break; + case SNDRV_PCM_FORMAT_S24_LE: + dai_cfg1 |= DA7210_DAI_WORD_S24_LE; + break; + default: + return -EINVAL; + } + + da7210_write(codec, DA7210_DAI_CFG1, dai_cfg1); + + /* FIXME + * + * It support 48K only now + */ + switch (params_rate(params)) { + case 48000: + if (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) { + reg = DA7210_DAC_HPF; + mask = ~DA7210_DAC_VOICE_EN; + } else { + reg = DA7210_ADC_HPF; + mask = ~DA7210_ADC_VOICE_EN; + } + break; + default: + return -EINVAL; + } + + value = da7210_read_reg_cache(codec, reg); + value &= mask; + da7210_write(codec, reg, value); + + return 0; +} + +/* + * Set DAI mode and Format + */ +static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + unsigned int dai_cfg1; + unsigned int dai_cfg3; + + dai_cfg1 = 0x7f & da7210_read(codec, DA7210_DAI_CFG1); + dai_cfg3 = 0xfc & da7210_read(codec, DA7210_DAI_CFG3); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + dai_cfg1 |= DA7210_DAI_MODE_MASTER; + break; + default: + return -EINVAL; + } + + /* FIXME + * + * It support I2S only now + */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + dai_cfg3 |= DA7210_DAI_FORMAT_I2SMODE; + break; + default: + return -EINVAL; + } + + /* FIXME + * + * It support 64bit data transmission only now + */ + dai_cfg1 |= DA7210_DAI_FLEN_64BIT; + + da7210_write(codec, DA7210_DAI_CFG1, dai_cfg1); + da7210_write(codec, DA7210_DAI_CFG3, dai_cfg3); + + return 0; +} + +#define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +/* DAI operations */ +static struct snd_soc_dai_ops da7210_dai_ops = { + .hw_params = da7210_hw_params, + .set_fmt = da7210_set_dai_fmt, +}; + +struct snd_soc_dai da7210_dai = { + .name = "DA7210 IIS", + .id = 0, + /* playback capabilities */ + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA7210_FORMATS, + }, + /* capture capabilities */ + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA7210_FORMATS, + }, + .ops = &da7210_dai_ops, +}; +EXPORT_SYMBOL_GPL(da7210_dai); + +/* + * Initialize the DA7210 driver + * register the mixer and dsp interfaces with the kernel + */ +static int da7210_init(struct da7210_priv *da7210) +{ + struct snd_soc_codec *codec = &da7210->codec; + int ret = 0; + + if (da7210_codec) { + dev_err(codec->dev, "Another da7210 is registered\n"); + return -EINVAL; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = da7210; + codec->name = "DA7210"; + codec->owner = THIS_MODULE; + codec->read = da7210_read; + codec->write = da7210_write; + codec->dai = &da7210_dai; + codec->num_dai = 1; + codec->hw_write = (hw_write_t)i2c_master_send; + codec->reg_cache_size = ARRAY_SIZE(da7210_reg); + codec->reg_cache = kmemdup(da7210_reg, + sizeof(da7210_reg), GFP_KERNEL); + + if (!codec->reg_cache) + return -ENOMEM; + + da7210_dai.dev = codec->dev; + da7210_codec = codec; + + ret = snd_soc_register_dai(&da7210_dai); + if (ret) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + return -ENOMEM; + } + + /* + * ADC settings + */ + + /* Enable Left & Right MIC PGA and Mic Bias */ + da7210_write(codec, DA7210_MIC_L, DA7210_MIC_L_EN | DA7210_MICBIAS_EN); + da7210_write(codec, DA7210_MIC_R, DA7210_MIC_R_EN); + + /* Enable Left and Right input PGA */ + da7210_write(codec, DA7210_INMIX_L, DA7210_IN_L_EN); + da7210_write(codec, DA7210_INMIX_R, DA7210_IN_R_EN); + + /* Enable ADC Highpass Filter */ + da7210_write(codec, DA7210_ADC_HPF, DA7210_ADC_HPF_EN); + + /* Enable Left and Right ADC */ + da7210_write(codec, DA7210_ADC, DA7210_ADC_L_EN | DA7210_ADC_R_EN); + + /* + * DAC settings + */ + + /* Enable DAC Highpass Filter */ + da7210_write(codec, DA7210_DAC_HPF, DA7210_DAC_HPF_EN); + + /* Enable Left and Right DAC */ + da7210_write(codec, DA7210_DAC_SEL, + DA7210_DAC_L_SRC_DAI_L | DA7210_DAC_L_EN | + DA7210_DAC_R_SRC_DAI_R | DA7210_DAC_R_EN); + + /* Enable Left and Right out PGA */ + da7210_write(codec, DA7210_OUTMIX_L, DA7210_OUT_L_EN); + da7210_write(codec, DA7210_OUTMIX_R, DA7210_OUT_R_EN); + + /* Enable Left and Right HeadPhone PGA */ + da7210_write(codec, DA7210_HP_CFG, + DA7210_HP_2CAP_MODE | DA7210_HP_SENSE_EN | + DA7210_HP_L_EN | DA7210_HP_MODE | DA7210_HP_R_EN); + + /* set DAI source to Left and Right ADC */ + da7210_write(codec, DA7210_DAI_SRC_SEL, + DA7210_DAI_OUT_R_SRC | DA7210_DAI_OUT_L_SRC); + + /* Enable DAI */ + da7210_write(codec, DA7210_DAI_CFG3, DA7210_DAI_OE | DA7210_DAI_EN); + + /* Diable PLL and bypass it */ + da7210_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); + + /* Bypass PLL and set MCLK freq rang to 10-20MHz */ + da7210_write(codec, DA7210_PLL_DIV3, + DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); + + /* Enable standbymode */ + da7210_write(codec, DA7210_STARTUP2, + DA7210_LOUT1_L_STBY | DA7210_LOUT1_R_STBY | + DA7210_LOUT2_STBY | DA7210_HP_L_STBY | + DA7210_HP_R_STBY | DA7210_DAC_L_STBY | DA7210_DAC_R_STBY); + da7210_write(codec, DA7210_STARTUP3, + DA7210_LIN1_L_STBY | DA7210_LIN1_R_STBY | + DA7210_LIN2_STBY | DA7210_MIC_L_STBY | + DA7210_MIC_R_STBY | DA7210_ADC_L_STBY | DA7210_ADC_R_STBY); + + /* Activate all enabled subsystem */ + da7210_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); + + return ret; +} + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static int da7210_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct da7210_priv *da7210; + struct snd_soc_codec *codec; + int ret; + + da7210 = kzalloc(sizeof(struct da7210_priv), GFP_KERNEL); + if (!da7210) + return -ENOMEM; + + codec = &da7210->codec; + codec->dev = &i2c->dev; + + i2c_set_clientdata(i2c, da7210); + codec->control_data = i2c; + + ret = da7210_init(da7210); + if (ret < 0) + pr_err("Failed to initialise da7210 audio codec\n"); + + return ret; +} + +static int da7210_i2c_remove(struct i2c_client *client) +{ + struct da7210_priv *da7210 = i2c_get_clientdata(client); + + snd_soc_unregister_dai(&da7210_dai); + kfree(da7210->codec.reg_cache); + kfree(da7210); + da7210_codec = NULL; + + return 0; +} + +static const struct i2c_device_id da7210_i2c_id[] = { + { "da7210", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, da7210_i2c_id); + +/* I2C codec control layer */ +static struct i2c_driver da7210_i2c_driver = { + .driver = { + .name = "DA7210 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = da7210_i2c_probe, + .remove = __devexit_p(da7210_i2c_remove), + .id_table = da7210_i2c_id, +}; +#endif + +static int da7210_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret; + + if (!da7210_codec) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = da7210_codec; + codec = da7210_codec; + + /* Register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) + goto pcm_err; + + dev_info(&pdev->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); + + /* Add the dapm controls */ + snd_soc_add_controls(codec, da7210_snd_controls, + ARRAY_SIZE(da7210_snd_controls)); + da7210_add_widgets(codec); + +pcm_err: + return ret; +} + +static int da7210_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_da7210 = { + .probe = da7210_probe, + .remove = da7210_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_da7210); + +static int __init da7210_modinit(void) +{ + int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&da7210_i2c_driver); +#endif + return ret; +} +module_init(da7210_modinit); + +static void __exit da7210_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&da7210_i2c_driver); +#endif +} +module_exit(da7210_exit); + +MODULE_DESCRIPTION("ASoC DA7210 driver"); +MODULE_AUTHOR("David Chen, Kuninori Morimoto"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/da7210.h b/sound/soc/codecs/da7210.h new file mode 100644 index 0000000..390d621 --- /dev/null +++ b/sound/soc/codecs/da7210.h @@ -0,0 +1,24 @@ +/* + * da7210.h -- audio driver for da7210 + * + * Copyright (c) 2009 Dialog Semiconductor + * Written by David Chen <Dajun.chen(a)diasemi.com> + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Cleanups by Kuninori Morimoto <morimoto.kuninori(a)renesas.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; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef _DA7210_H +#define _DA7210_H + +extern struct snd_soc_dai da7210_dai; +extern struct snd_soc_codec_device soc_codec_dev_da7210; + +#endif + -- 1.6.3.3

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Re: [alsa-devel] [PATCH] snd-usb-us122l: add product IDs of US-122MKII and US-144MKII
by Tobias Hansen 10 Dec '09

10 Dec '09

Alberto Martinez de la Ossa wrote: > Thanks, but.. > Applied where? I have checked the new sources at > http://ftp.kernel.org/pub/linux/kernel/people/tiwai/alsa/alsa-driver/ > And they remain the same with respect to the changes that Tobias made. > Could you please post the fixed patch, please? I also got errors when > trying to apply it. > > Regards, > Alberto Sorry, it somehow broke when I copied it into the mail. I'll better attach it now. Several people already reported that, for both US-122MKII and US-144MKII, adding the Product IDs doesn't make them work. But you can give it a try nevertheless and report back whether you had more luck. Regards, Tobias Hansen

1 0

[alsa-devel] [PATCH] opti93x: fix irq releasing if the irq cannot be allocated
by Krzysztof Helt 10 Dec '09

10 Dec '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Use the chip->irq to check if the irq should be released so the irq is not released if it has not been allocated. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/opti9xx/opti92x-ad1848.c | 18 ++++++++++-------- 1 files changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 8c88401..d8eac3f 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -548,10 +548,13 @@ __skip_mpu: static irqreturn_t snd_opti93x_interrupt(int irq, void *dev_id) { - struct snd_wss *codec = dev_id; - struct snd_opti9xx *chip = codec->card->private_data; + struct snd_opti9xx *chip = dev_id; + struct snd_wss *codec = chip->codec; unsigned char status; + if (!codec) + return IRQ_HANDLED; + status = snd_opti9xx_read(chip, OPTi9XX_MC_REG(11)); if ((status & OPTi93X_IRQ_PLAYBACK) && codec->playback_substream) snd_pcm_period_elapsed(codec->playback_substream); @@ -691,10 +694,9 @@ static void snd_card_opti9xx_free(struct snd_card *card) if (chip) { #ifdef OPTi93X - struct snd_wss *codec = chip->codec; - if (codec && codec->irq > 0) { - disable_irq(codec->irq); - free_irq(codec->irq, codec); + if (chip->irq > 0) { + disable_irq(chip->irq); + free_irq(chip->irq, chip); } release_and_free_resource(chip->res_mc_indir); #endif @@ -759,9 +761,9 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) #endif #ifdef OPTi93X error = request_irq(irq, snd_opti93x_interrupt, - IRQF_DISABLED, DEV_NAME" - WSS", codec); + IRQF_DISABLED, DEV_NAME" - WSS", chip); if (error < 0) { - snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", chip->irq); + snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", irq); return error; } #endif -- 1.6.4 ---------------------------------------------------------------------- Internetowi nie placa! Otworz Konto Direct. http://link.interia.pl/f24f7

2 1

[alsa-devel] [PATCH] dt019x: reuse als100 code
by Krzysztof Helt 10 Dec '09

10 Dec '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The als100 driver is so similar to the dt019x/als007 driver that one driver's source can be used for both drivers with only few ifdefs. Reuse the als100 code as the dt019x/als007 driver. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/als100.c | 86 +++++++++++---- sound/isa/dt019x.c | 323 +--------------------------------------------------- 2 files changed, 68 insertions(+), 341 deletions(-) diff --git a/sound/isa/als100.c b/sound/isa/als100.c index 5fd52e4..0f2e8d2 100644 --- a/sound/isa/als100.c +++ b/sound/isa/als100.c @@ -2,9 +2,13 @@ /* card-als100.c - driver for Avance Logic ALS100 based soundcards. Copyright (C) 1999-2000 by Massimo Piccioni <dafastidio(a)libero.it> + Copyright (C) 1999-2002 by Massimo Piccioni <dafastidio(a)libero.it> Thanks to Pierfrancesco 'qM2' Passerini. + Generalised for soundcards based on DT-0196 and ALS-007 chips + by Jonathan Woithe <jwoithe(a)physics.adelaide.edu.au>: June 2002. + 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 @@ -31,11 +35,16 @@ #include <sound/opl3.h> #include <sound/sb.h> +#ifdef DT019x +#define CARDNAME "dt019x" +#define PFX "dt019x: " +MODULE_DESCRIPTION("Diamond Technologies DT-019X / Avance Logic ALS-007"); +MODULE_SUPPORTED_DEVICE("{{Diamond Technologies DT-019X}," + "{Avance Logic ALS-007}}"); +#else +#define CARDNAME "als100" #define PFX "als100: " - -MODULE_AUTHOR("Massimo Piccioni <dafastidio(a)libero.it>"); MODULE_DESCRIPTION("Avance Logic ALS1X0"); -MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Avance Logic,ALS100 - PRO16PNP}," "{Avance Logic,ALS110}," "{Avance Logic,ALS120}," @@ -44,33 +53,47 @@ MODULE_SUPPORTED_DEVICE("{{Avance Logic,ALS100 - PRO16PNP}," "{Digimate,3D Sound}," "{Avance Logic,ALS120}," "{RTL,RTL3000}}"); +#endif + +MODULE_AUTHOR("Massimo Piccioni <dafastidio(a)libero.it>"); +MODULE_LICENSE("GPL"); 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 int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* PnP setup */ static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* PnP setup */ static int dma8[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* PnP setup */ +#ifndef DT019x static int dma16[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* PnP setup */ +#endif module_param_array(index, int, NULL, 0444); -MODULE_PARM_DESC(index, "Index value for als100 based soundcard."); +MODULE_PARM_DESC(index, "Index value for " CARDNAME " based soundcard."); module_param_array(id, charp, NULL, 0444); -MODULE_PARM_DESC(id, "ID string for als100 based soundcard."); +MODULE_PARM_DESC(id, "ID string for " CARDNAME " based soundcard."); module_param_array(enable, bool, NULL, 0444); -MODULE_PARM_DESC(enable, "Enable als100 based soundcard."); +MODULE_PARM_DESC(enable, "Enable " CARDNAME " based soundcard."); struct snd_card_als100 { - int dev_no; struct pnp_dev *dev; struct pnp_dev *devmpu; struct pnp_dev *devopl; struct snd_sb *chip; }; +#ifdef DT019x +static struct pnp_card_device_id snd_als100_pnpids[] = { + /* DT197A30 */ + { .id = "RWB1688", .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" }, } }, + /* DT0196 / ALS-007 */ + { .id = "ALS0007", .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" }, } }, + { .id = "", } /* end */ +}; +#else /* ALS100 devices */ static struct pnp_card_device_id snd_als100_pnpids[] = { /* ALS100 - PRO16PNP */ { .id = "ALS0001", .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" } } }, @@ -86,11 +109,10 @@ static struct pnp_card_device_id snd_als100_pnpids[] = { { .id = "RTL3000", .devs = { { "@@@2001" }, { "@X@2001" }, { "@H@2001" } } }, { .id = "", } /* end */ }; +#endif MODULE_DEVICE_TABLE(pnp_card, snd_als100_pnpids); -#define DRIVER_NAME "snd-card-als100" - static int __devinit snd_card_als100_pnp(int dev, struct snd_card_als100 *acard, struct pnp_card_link *card, const struct pnp_card_device_id *id) @@ -113,8 +135,12 @@ static int __devinit snd_card_als100_pnp(int dev, struct snd_card_als100 *acard, return err; } port[dev] = pnp_port_start(pdev, 0); +#ifdef DT019x + dma8[dev] = pnp_dma(pdev, 0); +#else dma8[dev] = pnp_dma(pdev, 1); dma16[dev] = pnp_dma(pdev, 0); +#endif irq[dev] = pnp_irq(pdev, 0); pdev = acard->devmpu; @@ -175,22 +201,34 @@ static int __devinit snd_card_als100_probe(int dev, } snd_card_set_dev(card, &pcard->card->dev); - if ((error = snd_sbdsp_create(card, port[dev], - irq[dev], - snd_sb16dsp_interrupt, - dma8[dev], - dma16[dev], - SB_HW_ALS100, &chip)) < 0) { + error = snd_sbdsp_create(card, port[dev], irq[dev], + snd_sb16dsp_interrupt, + dma8[dev], +#ifdef DT019x + -1, SB_HW_DT019X, +#else + dma16[dev], SB_HW_ALS100, +#endif + &chip); + if (error < 0) { snd_card_free(card); return error; } acard->chip = chip; +#ifdef DT019x + strcpy(card->driver, "DT-019X"); + strcpy(card->shortname, "Diamond Tech. DT-019X"); + sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d", + card->shortname, chip->name, chip->port, + irq[dev], dma8[dev]); +#else strcpy(card->driver, "ALS100"); strcpy(card->shortname, "Avance Logic ALS100"); sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d&%d", card->shortname, chip->name, chip->port, irq[dev], dma8[dev], dma16[dev]); +#endif if ((error = snd_sb16dsp_pcm(chip, 0, NULL)) < 0) { snd_card_free(card); @@ -203,9 +241,17 @@ static int __devinit snd_card_als100_probe(int dev, } if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) { - if (snd_mpu401_uart_new(card, 0, MPU401_HW_ALS100, + if (mpu_irq[dev] == SNDRV_AUTO_IRQ) + mpu_irq[dev] = -1; + if (snd_mpu401_uart_new(card, 0, +#ifdef DT019x + MPU401_HW_MPU401, +#else + MPU401_HW_ALS100, +#endif mpu_port[dev], 0, - mpu_irq[dev], IRQF_DISABLED, + mpu_irq[dev], + mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0, NULL) < 0) snd_printk(KERN_ERR PFX "no MPU-401 device at 0x%lx\n", mpu_port[dev]); } @@ -291,7 +337,7 @@ static int snd_als100_pnp_resume(struct pnp_card_link *pcard) static struct pnp_card_driver als100_pnpc_driver = { .flags = PNP_DRIVER_RES_DISABLE, - .name = "als100", + .name = CARDNAME, .id_table = snd_als100_pnpids, .probe = snd_als100_pnp_detect, .remove = __devexit_p(snd_als100_pnp_remove), @@ -312,7 +358,7 @@ static int __init alsa_card_als100_init(void) if (!als100_devices) { pnp_unregister_card_driver(&als100_pnpc_driver); #ifdef MODULE - snd_printk(KERN_ERR "no ALS100 based soundcards found\n"); + snd_printk(KERN_ERR "no " CARDNAME " based soundcards found\n"); #endif return -ENODEV; } diff --git a/sound/isa/dt019x.c b/sound/isa/dt019x.c index 80f5b1a..5fee291 100644 --- a/sound/isa/dt019x.c +++ b/sound/isa/dt019x.c @@ -1,321 +1,2 @@ - -/* - dt019x.c - driver for Diamond Technologies DT-0197H based soundcards. - Copyright (C) 1999, 2002 by Massimo Piccioni <dafastidio(a)libero.it> - - Generalised for soundcards based on DT-0196 and ALS-007 chips - by Jonathan Woithe <jwoithe(a)physics.adelaide.edu.au>: June 2002. - - 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/wait.h> -#include <linux/pnp.h> -#include <linux/moduleparam.h> -#include <sound/core.h> -#include <sound/initval.h> -#include <sound/mpu401.h> -#include <sound/opl3.h> -#include <sound/sb.h> - -#define PFX "dt019x: " - -MODULE_AUTHOR("Massimo Piccioni <dafastidio(a)libero.it>"); -MODULE_DESCRIPTION("Diamond Technologies DT-019X / Avance Logic ALS-007"); -MODULE_LICENSE("GPL"); -MODULE_SUPPORTED_DEVICE("{{Diamond Technologies DT-019X}," - "{Avance Logic ALS-007}}"); - -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; /* Enable this card */ -static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ -static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ -static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ -static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* PnP setup */ -static int mpu_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 DT-019X based soundcard."); -module_param_array(id, charp, NULL, 0444); -MODULE_PARM_DESC(id, "ID string for DT-019X based soundcard."); -module_param_array(enable, bool, NULL, 0444); -MODULE_PARM_DESC(enable, "Enable DT-019X based soundcard."); - -struct snd_card_dt019x { - struct pnp_dev *dev; - struct pnp_dev *devmpu; - struct pnp_dev *devopl; - struct snd_sb *chip; -}; - -static struct pnp_card_device_id snd_dt019x_pnpids[] = { - /* DT197A30 */ - { .id = "RWB1688", .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" }, } }, - /* DT0196 / ALS-007 */ - { .id = "ALS0007", .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" }, } }, - { .id = "", } -}; - -MODULE_DEVICE_TABLE(pnp_card, snd_dt019x_pnpids); - - -#define DRIVER_NAME "snd-card-dt019x" - - -static int __devinit snd_card_dt019x_pnp(int dev, struct snd_card_dt019x *acard, - struct pnp_card_link *card, - const struct pnp_card_device_id *pid) -{ - struct pnp_dev *pdev; - int err; - - acard->dev = pnp_request_card_device(card, pid->devs[0].id, NULL); - if (acard->dev == NULL) - return -ENODEV; - - acard->devmpu = pnp_request_card_device(card, pid->devs[1].id, NULL); - acard->devopl = pnp_request_card_device(card, pid->devs[2].id, NULL); - - pdev = acard->dev; - - err = pnp_activate_dev(pdev); - if (err < 0) { - snd_printk(KERN_ERR PFX "DT-019X 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); - snd_printdd("dt019x: found audio interface: port=0x%lx, irq=0x%x, dma=0x%x\n", - port[dev],irq[dev],dma8[dev]); - - pdev = acard->devmpu; - if (pdev != NULL) { - err = pnp_activate_dev(pdev); - if (err < 0) { - pnp_release_card_device(pdev); - snd_printk(KERN_ERR PFX "DT-019X MPU401 pnp configure failure, skipping\n"); - goto __mpu_error; - } - mpu_port[dev] = pnp_port_start(pdev, 0); - mpu_irq[dev] = pnp_irq(pdev, 0); - snd_printdd("dt019x: found MPU-401: port=0x%lx, irq=0x%x\n", - mpu_port[dev],mpu_irq[dev]); - } else { - __mpu_error: - acard->devmpu = NULL; - mpu_port[dev] = -1; - } - - pdev = acard->devopl; - if (pdev != NULL) { - err = pnp_activate_dev(pdev); - if (err < 0) { - pnp_release_card_device(pdev); - snd_printk(KERN_ERR PFX "DT-019X OPL3 pnp configure failure, skipping\n"); - goto __fm_error; - } - fm_port[dev] = pnp_port_start(pdev, 0); - snd_printdd("dt019x: found OPL3 synth: port=0x%lx\n",fm_port[dev]); - } else { - __fm_error: - acard->devopl = NULL; - fm_port[dev] = -1; - } - - return 0; -} - -static int __devinit snd_card_dt019x_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_dt019x *acard; - struct snd_opl3 *opl3; - - error = snd_card_create(index[dev], id[dev], THIS_MODULE, - sizeof(struct snd_card_dt019x), &card); - if (error < 0) - return error; - acard = card->private_data; - - snd_card_set_dev(card, &pcard->card->dev); - if ((error = snd_card_dt019x_pnp(dev, acard, pcard, pid))) { - snd_card_free(card); - return error; - } - - if ((error = snd_sbdsp_create(card, port[dev], - irq[dev], - snd_sb16dsp_interrupt, - dma8[dev], - -1, - SB_HW_DT019X, - &chip)) < 0) { - snd_card_free(card); - return error; - } - acard->chip = chip; - - strcpy(card->driver, "DT-019X"); - strcpy(card->shortname, "Diamond Tech. DT-019X"); - sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d", - card->shortname, chip->name, chip->port, - irq[dev], dma8[dev]); - - if ((error = snd_sb16dsp_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 (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) { - if (mpu_irq[dev] == SNDRV_AUTO_IRQ) - mpu_irq[dev] = -1; - if (snd_mpu401_uart_new(card, 0, -/* MPU401_HW_SB,*/ - MPU401_HW_MPU401, - mpu_port[dev], 0, - mpu_irq[dev], - mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0, - NULL) < 0) - snd_printk(KERN_ERR PFX "no MPU-401 device at 0x%lx ?\n", mpu_port[dev]); - } - - if (fm_port[dev] > 0 && fm_port[dev] != SNDRV_AUTO_PORT) { - if (snd_opl3_create(card, - fm_port[dev], - fm_port[dev] + 2, - OPL3_HW_AUTO, 0, &opl3) < 0) { - snd_printk(KERN_ERR PFX "no OPL device at 0x%lx-0x%lx ?\n", - fm_port[dev], fm_port[dev] + 2); - } else { - if ((error = snd_opl3_timer_new(opl3, 0, 1)) < 0) { - snd_card_free(card); - return error; - } - if ((error = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) { - snd_card_free(card); - return error; - } - } - } - - 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 dt019x_devices; - -static int __devinit snd_dt019x_pnp_probe(struct pnp_card_link *card, - const struct pnp_card_device_id *pid) -{ - static int dev; - int res; - - for ( ; dev < SNDRV_CARDS; dev++) { - if (!enable[dev]) - continue; - res = snd_card_dt019x_probe(dev, card, pid); - if (res < 0) - return res; - dev++; - dt019x_devices++; - return 0; - } - return -ENODEV; -} - -static void __devexit snd_dt019x_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_dt019x_pnp_suspend(struct pnp_card_link *pcard, pm_message_t state) -{ - struct snd_card *card = pnp_get_card_drvdata(pcard); - struct snd_card_dt019x *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_dt019x_pnp_resume(struct pnp_card_link *pcard) -{ - struct snd_card *card = pnp_get_card_drvdata(pcard); - struct snd_card_dt019x *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 dt019x_pnpc_driver = { - .flags = PNP_DRIVER_RES_DISABLE, - .name = "dt019x", - .id_table = snd_dt019x_pnpids, - .probe = snd_dt019x_pnp_probe, - .remove = __devexit_p(snd_dt019x_pnp_remove), -#ifdef CONFIG_PM - .suspend = snd_dt019x_pnp_suspend, - .resume = snd_dt019x_pnp_resume, -#endif -}; - -static int __init alsa_card_dt019x_init(void) -{ - int err; - - err = pnp_register_card_driver(&dt019x_pnpc_driver); - if (err) - return err; - - if (!dt019x_devices) { - pnp_unregister_card_driver(&dt019x_pnpc_driver); -#ifdef MODULE - snd_printk(KERN_ERR "no DT-019X / ALS-007 based soundcards found\n"); -#endif - return -ENODEV; - } - return 0; -} - -static void __exit alsa_card_dt019x_exit(void) -{ - pnp_unregister_card_driver(&dt019x_pnpc_driver); -} - -module_init(alsa_card_dt019x_init) -module_exit(alsa_card_dt019x_exit) +#define DT019x +#include "als100.c" -- 1.6.4 ---------------------------------------------------------------------- Internetowi nie placa! Otworz Konto Direct. http://link.interia.pl/f24f7

2 1

[alsa-devel] [PATCH 2/2] ASoC: Add FSI-DA7210 sound support for SuperH
by Kuninori Morimoto 10 Dec '09

10 Dec '09

Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- sound/soc/sh/Kconfig | 8 +++ sound/soc/sh/Makefile | 2 + sound/soc/sh/fsi-da7210.c | 117 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 127 insertions(+), 0 deletions(-) create mode 100644 sound/soc/sh/fsi-da7210.c diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 9e69765..8072a6d 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -47,4 +47,12 @@ config SND_FSI_AK4642 This option enables generic sound support for the FSI - AK4642 unit +config SND_FSI_DA7210 + bool "FSI-DA7210 sound support" + depends on SND_SOC_SH4_FSI + select SND_SOC_DA7210 + help + This option enables generic sound support for the + FSI - DA7210 unit + endmenu diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile index a699787..1d0ec0a 100644 --- a/sound/soc/sh/Makefile +++ b/sound/soc/sh/Makefile @@ -13,6 +13,8 @@ obj-$(CONFIG_SND_SOC_SH4_FSI) += snd-soc-fsi.o ## boards snd-soc-sh7760-ac97-objs := sh7760-ac97.o snd-soc-fsi-ak4642-objs := fsi-ak4642.o +snd-soc-fsi-da7210-objs := fsi-da7210.o obj-$(CONFIG_SND_SH7760_AC97) += snd-soc-sh7760-ac97.o obj-$(CONFIG_SND_FSI_AK4642) += snd-soc-fsi-ak4642.o +obj-$(CONFIG_SND_FSI_DA7210) += snd-soc-fsi-da7210.o diff --git a/sound/soc/sh/fsi-da7210.c b/sound/soc/sh/fsi-da7210.c new file mode 100644 index 0000000..e25e964 --- /dev/null +++ b/sound/soc/sh/fsi-da7210.c @@ -0,0 +1,117 @@ +/* + * fsi-da7210.c + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.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; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/io.h> +#include <linux/i2c.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <sound/sh_fsi.h> +#include "../codecs/da7210.h" + +static int fsi_da7210_init(struct snd_soc_codec *codec) +{ + return snd_soc_dai_set_fmt(&da7210_dai, + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + + /* Use this function to add machine dapm controls */ + return 0; +} + +static struct snd_soc_dai_link fsi_da7210_dai = { + .name = "DA7210", + .stream_name = "DA7210", + .cpu_dai = &fsi_soc_dai[1], /* FSI B */ + .codec_dai = &da7210_dai, + .init = fsi_da7210_init, +}; + +static struct snd_soc_card fsi_soc_card = { + .name = "FSI", + .platform = &fsi_soc_platform, + .dai_link = &fsi_da7210_dai, + .num_links = 1, +}; + +static struct snd_soc_device fsi_da7210_snd_devdata = { + .card = &fsi_soc_card, + .codec_dev = &soc_codec_dev_da7210, +}; + +#define DA7210_BUS 0 +#define DA7210_ADR (0x34 >> 1) +static int da7210_add_i2c_device(void) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = DA7210_ADR; + strlcpy(info.type, "da7210", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(DA7210_BUS); + if (!adapter) { + printk(KERN_ERR "can't get i2c adapter\n"); + return -ENODEV; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + printk(KERN_ERR "can't add i2c device\n"); + return -ENODEV; + } + + return 0; +} + +static struct platform_device *fsi_da7210_snd_device; + +static int __init fsi_da7210_sound_init(void) +{ + + int ret; + + da7210_add_i2c_device(); + + fsi_da7210_snd_device = platform_device_alloc("soc-audio", -1); + if (!fsi_da7210_snd_device) + return -ENOMEM; + + platform_set_drvdata(fsi_da7210_snd_device, &fsi_da7210_snd_devdata); + fsi_da7210_snd_devdata.dev = &fsi_da7210_snd_device->dev; + ret = platform_device_add(fsi_da7210_snd_device); + if (ret) + platform_device_put(fsi_da7210_snd_device); + + return ret; +} + +static void __exit fsi_da7210_sound_exit(void) +{ + platform_device_unregister(fsi_da7210_snd_device); +} + +module_init(fsi_da7210_sound_init); +module_exit(fsi_da7210_sound_exit); + +/* Module information */ +MODULE_DESCRIPTION("ALSA SoC FSI DA2710"); +MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori(a)renesas.com>"); +MODULE_LICENSE("GPL"); -- 1.6.3.3

2 3

[alsa-devel] [PATCH] hda/realtek: quirk for D945GCLF2 mainboard
by David Santinoli 09 Dec '09

09 Dec '09

Quirk for the ALC662 found on the Intel D945GCLF2 (and possibly other) mainboards. Signed-off-by: David Santinoli <david(a)santinoli.com> --- diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7e8b17a..b26fa53 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -16801,6 +16801,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E), SND_PCI_QUIRK(0x1849, 0x3662, "ASROCK K10N78FullHD-hSLI R3.0", ALC662_3ST_6ch_DIG), + SND_PCI_QUIRK(0x8086, 0xd604, "Intel mobo", ALC662_3ST_2ch_DIG), SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x", ALC663_ASUS_H13), {}

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

09 Dec '09

The following changes since commit 43f0de8d0298e624e6c3bf2185b6003a59b331bd: Mark Brown (1): S3C64XX: Staticise platform data for PCM devices are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Roel Kluin (1): ASoC: Wrong variable returned on error sound/soc/imx/mx1_mx2-pcm.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-)

2 1

[alsa-devel] [PATCH] Add direct git link to grub hda-analyzer
by Alexey Fisher 09 Dec '09

09 Dec '09

Just to save some time, add direct git link to grub hda-analyzer Signed-off-by: Alexey Fisher <bug-track(a)fisher-privat.net> modified: Documentation/sound/alsa/HD-Audio.txt --- Documentation/sound/alsa/HD-Audio.txt | 6 ++++-- 1 files changed, 4 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index 7b8a5f9..6325bec 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -624,11 +624,13 @@ hda-verb. The program gives you an easy-to-use GUI stuff for showing the widget information and adjusting the amp values, as well as the proc-compatible output. -The hda-analyzer is a part of alsa.git repository in -alsa-project.org: +The hda-analyzer: - http://git.alsa-project.org/?p=alsa.git;a=tree;f=hda-analyzer +is a part of alsa.git repository in alsa-project.org: + +- git://git.alsa-project.org/alsa.git Codecgraph ~~~~~~~~~~ -- 1.6.3.3

2 1

[alsa-devel] [PATCH] ASoC: tpa6130a2: Add support for regulator framework
by Ilkka Koskinen 09 Dec '09

09 Dec '09

Take the regulator framework in use for managing the power sources Signed-off-by: Ilkka Koskinen <ilkka.koskinen(a)nokia.com> --- sound/soc/codecs/tpa6130a2.c | 87 +++++++++++++++++++++++++++++++++-------- 1 files changed, 70 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 6b650c1..0eb33d4 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -25,6 +25,7 @@ #include <linux/device.h> #include <linux/i2c.h> #include <linux/gpio.h> +#include <linux/regulator/consumer.h> #include <sound/tpa6130a2-plat.h> #include <sound/soc.h> #include <sound/soc-dapm.h> @@ -34,10 +35,17 @@ static struct i2c_client *tpa6130a2_client; +#define TPA6130A2_NUM_SUPPLIES 2 +static const char *tpa6130a2_supply_names[TPA6130A2_NUM_SUPPLIES] = { + "CPVSS", + "Vdd", +}; + /* 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]; int power_gpio; unsigned char power_state; }; @@ -106,10 +114,11 @@ static void tpa6130a2_initialize(void) tpa6130a2_i2c_write(i, data->regs[i]); } -static void tpa6130a2_power(int power) +static int tpa6130a2_power(int power) { struct tpa6130a2_data *data; u8 val; + int ret; BUG_ON(tpa6130a2_client == NULL); data = i2c_get_clientdata(tpa6130a2_client); @@ -117,11 +126,20 @@ static void tpa6130a2_power(int power) mutex_lock(&data->mutex); if (power) { /* Power on */ - if (data->power_gpio >= 0) { + if (data->power_gpio >= 0) gpio_set_value(data->power_gpio, 1); - data->power_state = 1; - tpa6130a2_initialize(); + + ret = regulator_bulk_enable(ARRAY_SIZE(data->supplies), + data->supplies); + if (ret != 0) { + dev_err(&tpa6130a2_client->dev, + "Failed to enable supplies: %d\n", ret); + goto exit; } + + data->power_state = 1; + tpa6130a2_initialize(); + /* Clear SWS */ val = tpa6130a2_read(TPA6130A2_REG_CONTROL); val &= ~TPA6130A2_SWS; @@ -131,13 +149,25 @@ static void tpa6130a2_power(int power) val = tpa6130a2_read(TPA6130A2_REG_CONTROL); val |= TPA6130A2_SWS; tpa6130a2_i2c_write(TPA6130A2_REG_CONTROL, val); + /* Power off */ - if (data->power_gpio >= 0) { + if (data->power_gpio >= 0) gpio_set_value(data->power_gpio, 0); - data->power_state = 0; + + ret = regulator_bulk_disable(ARRAY_SIZE(data->supplies), + data->supplies); + if (ret != 0) { + dev_err(&tpa6130a2_client->dev, + "Failed to disable supplies: %d\n", ret); + goto exit; } + + data->power_state = 0; } + +exit: mutex_unlock(&data->mutex); + return ret; } static int tpa6130a2_get_reg(struct snd_kcontrol *kcontrol, @@ -299,15 +329,17 @@ static int tpa6130a2_right_event(struct snd_soc_dapm_widget *w, static int tpa6130a2_supply_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + int ret = 0; + switch (event) { case SND_SOC_DAPM_POST_PMU: - tpa6130a2_power(1); + ret = tpa6130a2_power(1); break; case SND_SOC_DAPM_POST_PMD: - tpa6130a2_power(0); + ret = tpa6130a2_power(0); break; } - return 0; + return ret; } static const struct snd_soc_dapm_widget tpa6130a2_dapm_widgets[] = { @@ -352,7 +384,7 @@ static int tpa6130a2_probe(struct i2c_client *client, struct device *dev; struct tpa6130a2_data *data; struct tpa6130a2_platform_data *pdata; - int ret; + int i, ret; dev = &client->dev; @@ -387,15 +419,25 @@ static int tpa6130a2_probe(struct i2c_client *client, if (ret < 0) { dev_err(dev, "Failed to request power GPIO (%d)\n", data->power_gpio); - goto fail; + goto err_gpio; } gpio_direction_output(data->power_gpio, 0); - } else { - data->power_state = 1; - tpa6130a2_initialize(); } - tpa6130a2_power(1); + 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); + goto err_regulator; + } + + ret = tpa6130a2_power(1); + if (ret != 0) + goto err_power; + /* Read version */ ret = tpa6130a2_i2c_read(TPA6130A2_REG_VERSION) & @@ -404,10 +446,18 @@ static int tpa6130a2_probe(struct i2c_client *client, dev_warn(dev, "UNTESTED version detected (%d)\n", ret); /* Disable the chip */ - tpa6130a2_power(0); + ret = tpa6130a2_power(0); + if (ret != 0) + goto err_power; return 0; -fail: + +err_power: + regulator_bulk_free(ARRAY_SIZE(data->supplies), data->supplies); +err_regulator: + if (data->power_gpio >= 0) + gpio_free(data->power_gpio); +err_gpio: kfree(data); i2c_set_clientdata(tpa6130a2_client, NULL); tpa6130a2_client = NULL; @@ -423,6 +473,9 @@ static int 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); + kfree(data); tpa6130a2_client = NULL; -- 1.6.0.4

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[alsa-devel] [PATCH 6/7] ASoC: S3C64XX: Pass I2S base address from platform data
by jassisinghbrar＠gmail.com 09 Dec '09

09 Dec '09

From: Jassi Brar <jassi.brar(a)samsung.com> Instead of having I2Sv2 core to figure out the base address of I2Sv3 controller pass the address directly. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/s3c64xx-i2s.c | 10 +++++++++- 1 files changed, 9 insertions(+), 1 deletions(-) diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 751a1b6..9cf62c2 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -166,6 +166,13 @@ static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Unable to get I2S SFR address\n"); return -ENXIO; } + + if (!request_mem_region(res->start, resource_size(res), + "s3c64xx-i2s")) { + dev_err(&pdev->dev, "Unable to request SFR region\n"); + return -EBUSY; + } + i2s->dma_capture->dma_addr = res->start + S3C2412_IISRXD; i2s->dma_playback->dma_addr = res->start + S3C2412_IISTXD; @@ -191,7 +198,8 @@ static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev) clk_enable(i2s->iis_cclk); - ret = s3c_i2sv2_probe(pdev, dai, i2s, 0); + ret = s3c_i2sv2_probe(pdev, dai, i2s, + i2s->dma_playback->dma_addr - S3C2412_IISTXD); if (ret) goto err_clk; -- 1.6.2.5

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[alsa-devel] [PATCH 2/7] ASoC: S3C64XX: Compress and generalize the CPU driver
by jassisinghbrar＠gmail.com 09 Dec '09

09 Dec '09

From: Jassi Brar <jassi.brar(a)samsung.com> The driver can be 'generalized' a bit by not hardcoding '2'(the number of I2Sv3 controllers that the driver can handle) at many places, instead we define a macro for it. That makes it easier to increase number of controllers by changing the parameter at just one place, this will be useful when there is support for newer SoCs, which have the same controller, only more in number. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/s3c64xx-i2s.c | 114 ++++++++++++++------------------------- 1 files changed, 41 insertions(+), 73 deletions(-) diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 8feb029..93ed3aa 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -32,6 +32,11 @@ #include "s3c-dma.h" #include "s3c64xx-i2s.h" +/* The value should be set to maximum of the total number + * of I2Sv3 controllers that any supported SoC has. + */ +#define MAX_I2SV3 2 + static struct s3c2410_dma_client s3c64xx_dma_client_out = { .name = "I2S PCM Stereo out" }; @@ -40,37 +45,12 @@ static struct s3c2410_dma_client s3c64xx_dma_client_in = { .name = "I2S PCM Stereo in" }; -static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_out[2] = { - [0] = { - .channel = DMACH_I2S0_OUT, - .client = &s3c64xx_dma_client_out, - .dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISTXD, - .dma_size = 4, - }, - [1] = { - .channel = DMACH_I2S1_OUT, - .client = &s3c64xx_dma_client_out, - .dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISTXD, - .dma_size = 4, - }, -}; - -static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_in[2] = { - [0] = { - .channel = DMACH_I2S0_IN, - .client = &s3c64xx_dma_client_in, - .dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISRXD, - .dma_size = 4, - }, - [1] = { - .channel = DMACH_I2S1_IN, - .client = &s3c64xx_dma_client_in, - .dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISRXD, - .dma_size = 4, - }, -}; +static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_out[MAX_I2SV3]; +static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_in[MAX_I2SV3]; +static struct s3c_i2sv2_info s3c64xx_i2s[MAX_I2SV3]; -static struct s3c_i2sv2_info s3c64xx_i2s[2]; +struct snd_soc_dai s3c64xx_i2s_dai[MAX_I2SV3]; +EXPORT_SYMBOL_GPL(s3c64xx_i2s_dai); static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) { @@ -163,55 +143,13 @@ static struct snd_soc_dai_ops s3c64xx_i2s_dai_ops = { .set_sysclk = s3c64xx_i2s_set_sysclk, }; -struct snd_soc_dai s3c64xx_i2s_dai[] = { - { - .name = "s3c64xx-i2s", - .id = 0, - .probe = s3c64xx_i2s_probe, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = S3C64XX_I2S_RATES, - .formats = S3C64XX_I2S_FMTS, - }, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = S3C64XX_I2S_RATES, - .formats = S3C64XX_I2S_FMTS, - }, - .ops = &s3c64xx_i2s_dai_ops, - .symmetric_rates = 1, - }, - { - .name = "s3c64xx-i2s", - .id = 1, - .probe = s3c64xx_i2s_probe, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = S3C64XX_I2S_RATES, - .formats = S3C64XX_I2S_FMTS, - }, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = S3C64XX_I2S_RATES, - .formats = S3C64XX_I2S_FMTS, - }, - .ops = &s3c64xx_i2s_dai_ops, - .symmetric_rates = 1, - }, -}; -EXPORT_SYMBOL_GPL(s3c64xx_i2s_dai); - static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev) { struct s3c_i2sv2_info *i2s; struct snd_soc_dai *dai; int ret; - if (pdev->id >= ARRAY_SIZE(s3c64xx_i2s)) { + if (pdev->id >= MAX_I2SV3) { dev_err(&pdev->dev, "id %d out of range\n", pdev->id); return -EINVAL; } @@ -219,10 +157,40 @@ static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev) i2s = &s3c64xx_i2s[pdev->id]; dai = &s3c64xx_i2s_dai[pdev->id]; dai->dev = &pdev->dev; + dai->name = "s3c64xx-i2s"; + dai->id = pdev->id; + 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_i2s_probe; + dai->ops = &s3c64xx_i2s_dai_ops; i2s->dma_capture = &s3c64xx_i2s_pcm_stereo_in[pdev->id]; i2s->dma_playback = &s3c64xx_i2s_pcm_stereo_out[pdev->id]; + if (pdev->id == 0) { + i2s->dma_capture->channel = DMACH_I2S0_IN; + i2s->dma_capture->dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISRXD; + i2s->dma_playback->channel = DMACH_I2S0_OUT; + i2s->dma_playback->dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISTXD; + } else { + i2s->dma_capture->channel = DMACH_I2S1_IN; + i2s->dma_capture->dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISRXD; + i2s->dma_playback->channel = DMACH_I2S1_OUT; + i2s->dma_playback->dma_addr = S3C64XX_PA_IIS1 + 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"); -- 1.6.2.5

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[alsa-devel] ASoC: S3C64XX: Patches for reforming s3c64xx-i2s driver
by jassisinghbrar＠gmail.com 09 Dec '09

09 Dec '09

From: Jassi Brar <jassi.brar(a)samsung.com> Hello, I am going to submit some cleanup and reforming patches to enable the s3c64xx-i2s.c to become generic enough to handle I2S-v3 controllers of other SoCs as well. Testing for build after each patch has been success. Thanks.

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[alsa-devel] [PATCH 5/7] ASoC: S3C64XX: Move gpio pin setup in platform init code
by jassisinghbrar＠gmail.com 09 Dec '09

09 Dec '09

From: Jassi Brar <jassi.brar(a)samsung.com> Like for other IPs, the configuration of MUX'ed GPIO to I2S mode should be done in platform initialization code. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/s3c64xx-i2s.c | 40 +++++++++----------------------------- 1 files changed, 10 insertions(+), 30 deletions(-) diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 8ca3c1d..751a1b6 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -16,17 +16,13 @@ #include <linux/module.h> #include <linux/device.h> #include <linux/clk.h> -#include <linux/gpio.h> #include <linux/io.h> #include <sound/soc.h> #include <plat/regs-s3c2412-iis.h> -#include <plat/gpio-bank-d.h> -#include <plat/gpio-bank-e.h> -#include <plat/gpio-cfg.h> +#include <plat/audio.h> -#include <mach/map.h> #include <mach/dma.h> #include "s3c-dma.h" @@ -106,30 +102,6 @@ struct clk *s3c64xx_i2s_get_clock(struct snd_soc_dai *dai) } EXPORT_SYMBOL_GPL(s3c64xx_i2s_get_clock); -static int s3c64xx_i2s_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - /* configure GPIO for i2s port */ - switch (dai->id) { - case 0: - s3c_gpio_cfgpin(S3C64XX_GPD(0), S3C64XX_GPD0_I2S0_CLK); - s3c_gpio_cfgpin(S3C64XX_GPD(1), S3C64XX_GPD1_I2S0_CDCLK); - s3c_gpio_cfgpin(S3C64XX_GPD(2), S3C64XX_GPD2_I2S0_LRCLK); - s3c_gpio_cfgpin(S3C64XX_GPD(3), S3C64XX_GPD3_I2S0_DI); - s3c_gpio_cfgpin(S3C64XX_GPD(4), S3C64XX_GPD4_I2S0_D0); - break; - case 1: - s3c_gpio_cfgpin(S3C64XX_GPE(0), S3C64XX_GPE0_I2S1_CLK); - s3c_gpio_cfgpin(S3C64XX_GPE(1), S3C64XX_GPE1_I2S1_CDCLK); - s3c_gpio_cfgpin(S3C64XX_GPE(2), S3C64XX_GPE2_I2S1_LRCLK); - s3c_gpio_cfgpin(S3C64XX_GPE(3), S3C64XX_GPE3_I2S1_DI); - s3c_gpio_cfgpin(S3C64XX_GPE(4), S3C64XX_GPE4_I2S1_D0); - } - - return 0; -} - - #define S3C64XX_I2S_RATES \ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ @@ -145,6 +117,7 @@ static struct snd_soc_dai_ops s3c64xx_i2s_dai_ops = { static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev) { + struct s3c_audio_pdata *i2s_pdata; struct s3c_i2sv2_info *i2s; struct snd_soc_dai *dai; struct resource *res; @@ -169,7 +142,6 @@ static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev) dai->capture.channels_max = 2; dai->capture.rates = S3C64XX_I2S_RATES; dai->capture.formats = S3C64XX_I2S_FMTS; - dai->probe = s3c64xx_i2s_probe; dai->ops = &s3c64xx_i2s_dai_ops; i2s->dma_capture = &s3c64xx_i2s_pcm_stereo_in[pdev->id]; @@ -202,6 +174,14 @@ static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev) i2s->dma_playback->client = &s3c64xx_dma_client_out; i2s->dma_playback->dma_size = 4; + i2s_pdata = pdev->dev.platform_data; + + /* Configure the I2S pins if MUX'ed */ + if (i2s_pdata && i2s_pdata->cfg_gpio && i2s_pdata->cfg_gpio(pdev)) { + dev_err(&pdev->dev, "Unable to configure gpio\n"); + return -EINVAL; + } + 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"); -- 1.6.2.5

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[alsa-devel] [PATCH 7/7] ASoC: S3C24XX: Remove I2Sv3 specific check
by jassisinghbrar＠gmail.com 09 Dec '09

09 Dec '09

From: Jassi Brar <jassi.brar(a)samsung.com> Since, now the I2Sv3 driver itself figures out and provides the start address of the base register, the s3c64xx specific patch can be discarded in the I2Sv2 core code. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/s3c-i2s-v2.c | 18 ------------------ 1 files changed, 0 insertions(+), 18 deletions(-) diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index e994d83..422b74c 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -628,24 +628,6 @@ int s3c_i2sv2_probe(struct platform_device *pdev, /* record our i2s structure for later use in the callbacks */ dai->private_data = i2s; - if (!base) { - struct resource *res = platform_get_resource(pdev, - IORESOURCE_MEM, - 0); - if (!res) { - dev_err(dev, "Unable to get register resource\n"); - return -ENXIO; - } - - if (!request_mem_region(res->start, resource_size(res), - "s3c64xx-i2s-v4")) { - dev_err(dev, "Unable to request register region\n"); - return -EBUSY; - } - - base = res->start; - } - i2s->regs = ioremap(base, 0x100); if (i2s->regs == NULL) { dev_err(dev, "cannot ioremap registers\n"); -- 1.6.2.5

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[alsa-devel] How to use soc API without codec driver
by Marc Garnier 09 Dec '09

09 Dec '09

Hi, I need to receive and transmit PCM data through SSC port of a sam9261 to a device which doesn't have any kind of I/O controls (neither i2c nor spi). So, I wonder how to deal with the codec driver side. Someone could explain me what are the minimum requirements to write a Asoc driver of this kind?

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[alsa-devel] [PATCH 3/7] ARM: S3C64XX: Add I2S resources in platform code
by jassisinghbrar＠gmail.com 09 Dec '09

09 Dec '09

From: Jassi Brar <jassi.brar(a)samsung.com> Define resources needed by the I2S drivers in platform code. Audio cfg_gpio callback and IORESOURCE_DMA are added. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- arch/arm/plat-s3c64xx/dev-audio.c | 91 ++++++++++++++++++++++++++++++++++++- 1 files changed, 90 insertions(+), 1 deletions(-) diff --git a/arch/arm/plat-s3c64xx/dev-audio.c b/arch/arm/plat-s3c64xx/dev-audio.c index a21a88f..cdba3a7 100644 --- a/arch/arm/plat-s3c64xx/dev-audio.c +++ b/arch/arm/plat-s3c64xx/dev-audio.c @@ -3,7 +3,6 @@ * Copyright 2009 Wolfson Microelectronics * Mark Brown <broonie(a)opensource.wolfsonmicro.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. @@ -20,16 +19,69 @@ #include <plat/devs.h> #include <plat/audio.h> +#include <plat/gpio-bank-c.h> #include <plat/gpio-bank-d.h> #include <plat/gpio-bank-e.h> +#include <plat/gpio-bank-h.h> #include <plat/gpio-cfg.h> +static int s3c64xx_i2sv3_cfg_gpio(struct platform_device *pdev) +{ + switch (pdev->id) { + case 0: + s3c_gpio_cfgpin(S3C64XX_GPD(0), S3C64XX_GPD0_I2S0_CLK); + s3c_gpio_cfgpin(S3C64XX_GPD(1), S3C64XX_GPD1_I2S0_CDCLK); + s3c_gpio_cfgpin(S3C64XX_GPD(2), S3C64XX_GPD2_I2S0_LRCLK); + s3c_gpio_cfgpin(S3C64XX_GPD(3), S3C64XX_GPD3_I2S0_DI); + s3c_gpio_cfgpin(S3C64XX_GPD(4), S3C64XX_GPD4_I2S0_D0); + break; + case 1: + s3c_gpio_cfgpin(S3C64XX_GPE(0), S3C64XX_GPE0_I2S1_CLK); + s3c_gpio_cfgpin(S3C64XX_GPE(1), S3C64XX_GPE1_I2S1_CDCLK); + s3c_gpio_cfgpin(S3C64XX_GPE(2), S3C64XX_GPE2_I2S1_LRCLK); + s3c_gpio_cfgpin(S3C64XX_GPE(3), S3C64XX_GPE3_I2S1_DI); + s3c_gpio_cfgpin(S3C64XX_GPE(4), S3C64XX_GPE4_I2S1_D0); + default: + printk(KERN_DEBUG "Invalid I2S Controller number!"); + return -EINVAL; + } + + return 0; +} + +static int s3c64xx_i2sv4_cfg_gpio(struct platform_device *pdev) +{ + s3c_gpio_cfgpin(S3C64XX_GPC(4), S3C64XX_GPC4_I2S0_V40_DO); + s3c_gpio_cfgpin(S3C64XX_GPC(5), S3C64XX_GPC5_I2S1_V40_DO); + s3c_gpio_cfgpin(S3C64XX_GPC(7), S3C64XX_GPC7_I2S2_V40_DO); + 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 struct resource s3c64xx_iis0_resource[] = { [0] = { .start = S3C64XX_PA_IIS0, .end = S3C64XX_PA_IIS0 + 0x100 - 1, .flags = IORESOURCE_MEM, }, + [1] = { + .start = DMACH_I2S0_OUT, + .end = DMACH_I2S0_OUT, + .flags = IORESOURCE_DMA, + }, + [2] = { + .start = DMACH_I2S0_IN, + .end = DMACH_I2S0_IN, + .flags = IORESOURCE_DMA, + }, +}; + +struct s3c_audio_pdata s3c_i2s0_pdata = { + .cfg_gpio = s3c64xx_i2sv3_cfg_gpio, }; struct platform_device s3c64xx_device_iis0 = { @@ -37,6 +89,9 @@ struct platform_device s3c64xx_device_iis0 = { .id = 0, .num_resources = ARRAY_SIZE(s3c64xx_iis0_resource), .resource = s3c64xx_iis0_resource, + .dev = { + .platform_data = &s3c_i2s0_pdata, + }, }; EXPORT_SYMBOL(s3c64xx_device_iis0); @@ -46,6 +101,20 @@ static struct resource s3c64xx_iis1_resource[] = { .end = S3C64XX_PA_IIS1 + 0x100 - 1, .flags = IORESOURCE_MEM, }, + [1] = { + .start = DMACH_I2S1_OUT, + .end = DMACH_I2S1_OUT, + .flags = IORESOURCE_DMA, + }, + [2] = { + .start = DMACH_I2S1_IN, + .end = DMACH_I2S1_IN, + .flags = IORESOURCE_DMA, + }, +}; + +struct s3c_audio_pdata s3c_i2s1_pdata = { + .cfg_gpio = s3c64xx_i2sv3_cfg_gpio, }; struct platform_device s3c64xx_device_iis1 = { @@ -53,6 +122,9 @@ struct platform_device s3c64xx_device_iis1 = { .id = 1, .num_resources = ARRAY_SIZE(s3c64xx_iis1_resource), .resource = s3c64xx_iis1_resource, + .dev = { + .platform_data = &s3c_i2s1_pdata, + }, }; EXPORT_SYMBOL(s3c64xx_device_iis1); @@ -62,6 +134,20 @@ static struct resource s3c64xx_iisv4_resource[] = { .end = S3C64XX_PA_IISV4 + 0x100 - 1, .flags = IORESOURCE_MEM, }, + [1] = { + .start = DMACH_HSI_I2SV40_TX, + .end = DMACH_HSI_I2SV40_TX, + .flags = IORESOURCE_DMA, + }, + [2] = { + .start = DMACH_HSI_I2SV40_RX, + .end = DMACH_HSI_I2SV40_RX, + .flags = IORESOURCE_DMA, + }, +}; + +struct s3c_audio_pdata s3c_i2sv4_pdata = { + .cfg_gpio = s3c64xx_i2sv4_cfg_gpio, }; struct platform_device s3c64xx_device_iisv4 = { @@ -69,6 +155,9 @@ struct platform_device s3c64xx_device_iisv4 = { .id = -1, .num_resources = ARRAY_SIZE(s3c64xx_iisv4_resource), .resource = s3c64xx_iisv4_resource, + .dev = { + .platform_data = &s3c_i2sv4_pdata, + }, }; EXPORT_SYMBOL(s3c64xx_device_iisv4); -- 1.6.2.5

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[alsa-devel] [PATCH 1/7] ASoC: S3C64XX: Remove unnecessary header includes
by jassisinghbrar＠gmail.com 09 Dec '09

09 Dec '09

From: Jassi Brar <jassi.brar(a)samsung.com> Removed redundant header includes which make no difference to compilation. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/s3c64xx-i2s.c | 6 ------ 1 files changed, 0 insertions(+), 6 deletions(-) diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index cc7edb5..8feb029 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -15,16 +15,10 @@ #include <linux/init.h> #include <linux/module.h> #include <linux/device.h> -#include <linux/delay.h> #include <linux/clk.h> -#include <linux/kernel.h> #include <linux/gpio.h> #include <linux/io.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> #include <sound/soc.h> #include <plat/regs-s3c2412-iis.h> -- 1.6.2.5

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Re: [alsa-devel] [PATCH v2] Introduce config for intel dg45id board
by Takashi Iwai 08 Dec '09

08 Dec '09

At Tue, 08 Dec 2009 17:30:18 +0100, Alexey Fisher wrote: > > Am Dienstag, den 08.12.2009, 14:15 +0100 schrieb Takashi Iwai: > > At Tue, 08 Dec 2009 14:02:43 +0100, > > Alexey Fisher wrote: > > > > > > Am Dienstag, den 08.12.2009, 12:04 +0100 schrieb Takashi Iwai: > > > > At Sun, 6 Dec 2009 11:29:13 +0100, > > > > Alexey Fisher wrote: > > > > > > > > > > This patch introduce pin config and some workarounds for dg45id board. > > > > > Currently tested Mic + Surround 7.1 on rear panel, and Mic + HP on front panel. > > > > > SPDIF front and SPDIF rear are untested. > > > > > Both Mics provide VREF_80 (4,05 V) in mic mode and no VREF in line-in mode. > > > > > > > > > > Signed-off-by: Alexey Fisher <bug-track(a)fisher-privat.net> > > > > > > > > Thanks for the patch. > > > > > > > > But, I still don't see the reason for so many init verbs, especially > > > > doing static routings. Can't be they connected properly by the > > > > parser? If so, it's the parser to be fixed, not a quirky init table. > > > > > > Ok. I prefer to have the part with mixer. The driver currently can't > > > handle this. > > > > What do you mean exactly with "mixer"? > > > > > It seems to make some problem with front HP. By default All > > > mixer inputs use 0x0a (Front HP out), in this situation i get hi freq > > > noise on 0x0a. So or driver should learn to work with mixer or...? > > > > Not sure what you are talking about here... > > I talking about widget like this: > > Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 > 0x80] > Connection: 5 > 0x18 0x19 0x1a 0x1b 0x1d > > this is an example from my laptop. Normally this widget connected to the > audio selector like this: > > Node 0x23 [Audio Selector] wcaps 0x300101: Stereo > Connection: 7 > 0x18 0x19 0x1a 0x1b 0x1d 0x12* 0x0b > > ALSA will list only attached to "Audio Selector" "Mic" or "Line in" > pins, not "Audio mixer". ALSA do not give control for audio mixer too. Just because STAC/IDT codecs have no mixer widget. It's a hardware issue. > > > > And, your machine has really no headphone detection? I mean, not > > > > about your taste but it's not physically doable? > > > > > > HP detection working on rear green (0x0d), not on front green (0x0a). > > > May be there is something wrong with connector. Anyway, it working fine > > > under M$. May be we should provide UI control for this (to make user > > > completely confused:)? > > > > If Windows driver can detect the front HP jack, it means the > > connection is alive. You can try hda-verb to check whether the > > pin-detection works, independently from the driver setup. > > Pin detection working only on rear panel, on front panel it do not > working and on windows and on linux. Probably broken front panel jack... > I assume windwos use different logic. OK, then it's likely a typical problem of the cabling and the case model. In many cases, an AC97-style case is connected to a HD-audio board. > linux logic: if no hp, play sound to speaker and mute hp. if hp detected > - unmute hp and mute speaker. > winodws logic: if no hp detected play to hp and to speaker, if hp > detected - mute speaker and continue play to hp. > > I think, if ALSA will behave in same way it will need less hp quirks. Yes, that's how no_jd model works. But, keeping the headphone output even with unplugged state means you are wasting unneeded power. The driver powers down the circuit appropriately if a pin isn't being used. That's why the headphone detection is implemented as default. And, whether the HP detection works or not doesn't depend on the mobo. It's rather the connection. So, giving the fixed "no-jd" option for a certain PCI SSID is basically wrong. Someone else might have a same mobo but with a right case with the front-panel jack detection. That is, it's fine to create a new quirk model, but assigning statically to that workaround isn't always acceptable. thanks, Takashi

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[alsa-devel] [Updated]iMac_91_sound_patch.
by Justin P. Mattock 08 Dec '09

08 Dec '09

This is an updated patch for the Apple iMac 9,1 model to add sound. Original patch posted here: http://article.gmane.org/gmane.linux.alsa.devel/61361/match= I have been using this patch for a while now and have to say it works vary well, except for a few minor things: With the iMac 24-inch 3.06GHz Intel Core 2 Duo everything seems to be working as it should, although I have not looked into the microphone (never really use one, nor have any apps to test, my guess is it doesn't work, or I never figured out how to get it to work). With the iMac 24-inch 2.66GHz Intel Core 2 Duo everything is the same as with the above machine except I'm hearing a light scratchy/distortion noise come out of the speakers when using headphones(above machine does not do this). Other than that the sound level is great(especially with good Dj headphones). Now onto this patch. I took the liberty of sighning off on it although I don't feel comfortable doing so, since this is not my work(except for a few changes to fit the latest kernel). In reading: http://article.gmane.org/gmane.linux.alsa.devel/61596/match= there was mention of a sign off, but something happened. Anyways I'll leave it up to the people in charge of patches to make the descision on what to do with this. Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_realtek.c | 111 ++++++++++++++++++++++++++ 2 files changed, 112 insertions(+), 0 deletions(-)

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[alsa-devel] [PATCH] snd-usb-us122l: add product IDs of US-122MKII and US-144MKII
by Tobias Hansen 08 Dec '09

08 Dec '09

snd-usb-us122l: add product IDs of US-122MKII and US-144MKII I added the product IDs of the new revisions of the devices, so owners can test whether this suffices to make them work. Patched against ALSA snapshot 20091207. Signed-off-by: Tobias Hansen <Tobias.Hansen at physik.uni-hamburg.de <http://mailman.alsa-project.org/mailman/listinfo/alsa-devel>> --- diff -upr alsa-driver.orig/alsa-kernel/usb/usx2y/us122l.c alsa-driver/alsa-kernel/usb/usx2y/us122l.c --- alsa-driver.orig/alsa-kernel/usb/usx2y/us122l.c 2009-12-07 00:05:40.000000000 +0100 +++ alsa-driver/alsa-kernel/usb/usx2y/us122l.c 2009-12-07 18:50:27.801975964 +0100 @@ -194,7 +194,8 @@ static int usb_stream_hwdep_open(struct if (!us122l->first) us122l->first = file; - if (us122l->dev->descriptor.idProduct == USB_ID_US144) { + if (us122l->dev->descriptor.idProduct == USB_ID_US144 || + us122l->dev->descriptor.idProduct == USB_ID_US144MKII) { iface = usb_ifnum_to_if(us122l->dev, 0); usb_autopm_get_interface(iface); } @@ -209,7 +210,8 @@ static int usb_stream_hwdep_release(stru struct usb_interface *iface; snd_printdd(KERN_DEBUG "%p %p\n", hw, file); - if (us122l->dev->descriptor.idProduct == USB_ID_US144) { + if (us122l->dev->descriptor.idProduct == USB_ID_US144 || + us122l->dev->descriptor.idProduct == USB_ID_US144MKII) { iface = usb_ifnum_to_if(us122l->dev, 0); usb_autopm_put_interface(iface); } @@ -476,7 +478,8 @@ static bool us122l_create_card(struct sn int err; struct us122l *us122l = US122L(card); - if (us122l->dev->descriptor.idProduct == USB_ID_US144) { + if (us122l->dev->descriptor.idProduct == USB_ID_US144 || + us122l->dev->descriptor.idProduct == USB_ID_US144MKII) { err = usb_set_interface(us122l->dev, 0, 1); if (err) { snd_printk(KERN_ERR "usb_set_interface error \n"); @@ -495,7 +498,8 @@ static bool us122l_create_card(struct sn if (!us122l_start(us122l, 44100, 256)) return false; - if (us122l->dev->descriptor.idProduct == USB_ID_US144) + if (us122l->dev->descriptor.idProduct == USB_ID_US144 || + us122l->dev->descriptor.idProduct == USB_ID_US144MKII) err = us144_create_usbmidi(card); else err = us122l_create_usbmidi(card); @@ -597,7 +601,8 @@ static int snd_us122l_probe(struct usb_i struct snd_card *card; int err; - if (device->descriptor.idProduct == USB_ID_US144 + if ((device->descriptor.idProduct == USB_ID_US144 || + us122l->dev->descriptor.idProduct == USB_ID_US144MKII) && device->speed == USB_SPEED_HIGH) { snd_printk(KERN_ERR "disable ehci-hcd to run US-144 \n"); return -ENODEV; @@ -692,7 +697,8 @@ static int snd_us122l_resume(struct usb_ mutex_lock(&us122l->mutex); /* needed, doesn't restart without: */ - if (us122l->dev->descriptor.idProduct == USB_ID_US144) { + if (us122l->dev->descriptor.idProduct == USB_ID_US144 || + us122l->dev->descriptor.idProduct == USB_ID_US144MKII) { err = usb_set_interface(us122l->dev, 0, 1); if (err) { snd_printk(KERN_ERR "usb_set_interface error \n"); @@ -737,6 +743,16 @@ static struct usb_device_id snd_us122l_u .idVendor = 0x0644, .idProduct = USB_ID_US144 }, + { + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = 0x0644, + .idProduct = USB_ID_US122MKII + }, + { + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = 0x0644, + .idProduct = USB_ID_US144MKII + }, { /* terminator */ } }; diff -upr alsa-driver.orig/alsa-kernel/usb/usx2y/us122l.h alsa-driver/alsa-kernel/usb/usx2y/us122l.h --- alsa-driver.orig/alsa-kernel/usb/usx2y/us122l.h 2009-12-07 00:05:40.000000000 +0100 +++ alsa-driver/alsa-kernel/usb/usx2y/us122l.h 2009-12-07 18:44:03.990236199 +0100 @@ -25,5 +25,7 @@ struct us122l { #define USB_ID_US122L 0x800E #define USB_ID_US144 0x800F +#define USB_ID_US122MKII 0x8021 +#define USB_ID_US144MKII 0x8020 #endif

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[alsa-devel] [PATCH] opti93x: set MC indirect registers base from PnP data
by Krzysztof Helt 08 Dec '09

08 Dec '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The PnP data on the OPTI931 and OPTI933 contains io port range for the MC indirect registers. Use the PnP range instead of hardwired value 0xE0E. Also, request region of MC indirect registers so it is marked as used to other drivers (this was missing previously). --- This is the second version of the patch. The request_region() is called in a single place now. sound/isa/opti9xx/opti92x-ad1848.c | 112 +++++++++++++++++++++-------------- 1 files changed, 67 insertions(+), 45 deletions(-) diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index d08c389..8c88401 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -135,6 +135,8 @@ struct snd_opti9xx { unsigned long mc_base_size; #ifdef OPTi93X unsigned long mc_indir_index; + unsigned long mc_indir_size; + struct resource *res_mc_indir; struct snd_wss *codec; #endif /* OPTi93X */ unsigned long pwd_reg; @@ -231,7 +233,10 @@ static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip, case OPTi9XX_HW_82C931: case OPTi9XX_HW_82C933: chip->mc_base = (hardware == OPTi9XX_HW_82C930) ? 0xf8f : 0xf8d; - chip->mc_indir_index = 0xe0e; + if (!chip->mc_indir_index) { + chip->mc_indir_index = 0xe0e; + chip->mc_indir_size = 2; + } chip->password = 0xe4; chip->pwd_reg = 0; break; @@ -560,57 +565,69 @@ static irqreturn_t snd_opti93x_interrupt(int irq, void *dev_id) #endif /* OPTi93X */ -static int __devinit snd_card_opti9xx_detect(struct snd_card *card, - struct snd_opti9xx *chip) +static int __devinit snd_opti9xx_read_check(struct snd_opti9xx *chip) { - int i, err; + unsigned char value; +#ifdef OPTi93X + unsigned long flags; +#endif + chip->res_mc_base = request_region(chip->mc_base, chip->mc_base_size, + "OPTi9xx MC"); + if (chip->res_mc_base == NULL) + return -EBUSY; #ifndef OPTi93X - for (i = OPTi9XX_HW_82C928; i < OPTi9XX_HW_82C930; i++) { - unsigned char value; + value = snd_opti9xx_read(chip, OPTi9XX_MC_REG(1)); + if (value != 0xff && value != inb(chip->mc_base + OPTi9XX_MC_REG(1))) + if (value == snd_opti9xx_read(chip, OPTi9XX_MC_REG(1))) + return 0; +#else /* OPTi93X */ + chip->res_mc_indir = request_region(chip->mc_indir_index, + chip->mc_indir_size, + "OPTi93x MC"); + if (chip->res_mc_indir == NULL) + return -EBUSY; - if ((err = snd_opti9xx_init(chip, i)) < 0) - return err; + spin_lock_irqsave(&chip->lock, flags); + outb(chip->password, chip->mc_base + chip->pwd_reg); + outb(((chip->mc_indir_index & 0x1f0) >> 4), chip->mc_base); + spin_unlock_irqrestore(&chip->lock, flags); - if ((chip->res_mc_base = request_region(chip->mc_base, chip->mc_base_size, "OPTi9xx MC")) == NULL) - continue; + value = snd_opti9xx_read(chip, OPTi9XX_MC_REG(7)); + snd_opti9xx_write(chip, OPTi9XX_MC_REG(7), 0xff - value); + if (snd_opti9xx_read(chip, OPTi9XX_MC_REG(7)) == 0xff - value) + return 0; - value = snd_opti9xx_read(chip, OPTi9XX_MC_REG(1)); - if ((value != 0xff) && (value != inb(chip->mc_base + 1))) - if (value == snd_opti9xx_read(chip, OPTi9XX_MC_REG(1))) - return 1; + release_and_free_resource(chip->res_mc_indir); + chip->res_mc_indir = NULL; +#endif /* OPTi93X */ + release_and_free_resource(chip->res_mc_base); + chip->res_mc_base = NULL; - release_and_free_resource(chip->res_mc_base); - chip->res_mc_base = NULL; + return -ENODEV; +} - } -#else /* OPTi93X */ - for (i = OPTi9XX_HW_82C931; i >= OPTi9XX_HW_82C930; i--) { - unsigned long flags; - unsigned char value; +static int __devinit snd_card_opti9xx_detect(struct snd_card *card, + struct snd_opti9xx *chip) +{ + int i, err; - if ((err = snd_opti9xx_init(chip, i)) < 0) +#ifndef OPTi93X + for (i = OPTi9XX_HW_82C928; i < OPTi9XX_HW_82C930; i++) { +#else + for (i = OPTi9XX_HW_82C931; i >= OPTi9XX_HW_82C930; i--) { +#endif + err = snd_opti9xx_init(chip, i); + if (err < 0) return err; - if ((chip->res_mc_base = request_region(chip->mc_base, chip->mc_base_size, "OPTi9xx MC")) == NULL) - continue; - - spin_lock_irqsave(&chip->lock, flags); - outb(chip->password, chip->mc_base + chip->pwd_reg); - outb(((chip->mc_indir_index & (1 << 8)) >> 4) | - ((chip->mc_indir_index & 0xf0) >> 4), chip->mc_base); - spin_unlock_irqrestore(&chip->lock, flags); - - value = snd_opti9xx_read(chip, OPTi9XX_MC_REG(7)); - snd_opti9xx_write(chip, OPTi9XX_MC_REG(7), 0xff - value); - if (snd_opti9xx_read(chip, OPTi9XX_MC_REG(7)) == 0xff - value) + err = snd_opti9xx_read_check(chip); + if (err == 0) return 1; - - release_and_free_resource(chip->res_mc_base); - chip->res_mc_base = NULL; +#ifdef OPTi93X + chip->mc_indir_index = 0; +#endif } -#endif /* OPTi93X */ - return -ENODEV; } @@ -639,6 +656,8 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip, #ifdef OPTi93X port = pnp_port_start(pdev, 0) - 4; fm_port = pnp_port_start(pdev, 1) + 8; + chip->mc_indir_index = pnp_port_start(pdev, 3) + 2; + chip->mc_indir_size = pnp_port_len(pdev, 3) - 2; #else if (pid->driver_data != 0x0924) port = pnp_port_start(pdev, 1); @@ -669,7 +688,7 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip, static void snd_card_opti9xx_free(struct snd_card *card) { struct snd_opti9xx *chip = card->private_data; - + if (chip) { #ifdef OPTi93X struct snd_wss *codec = chip->codec; @@ -677,6 +696,7 @@ static void snd_card_opti9xx_free(struct snd_card *card) disable_irq(codec->irq); free_irq(codec->irq, codec); } + release_and_free_resource(chip->res_mc_indir); #endif release_and_free_resource(chip->res_mc_base); } @@ -696,11 +716,6 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) struct snd_rawmidi *rmidi; struct snd_hwdep *synth; - if (! chip->res_mc_base && - (chip->res_mc_base = request_region(chip->mc_base, chip->mc_base_size, - "OPTi9xx MC")) == NULL) - return -ENOMEM; - #if defined(CS4231) || defined(OPTi93X) xdma2 = dma2; #else @@ -954,6 +969,13 @@ static int __devinit snd_opti9xx_pnp_probe(struct pnp_card_link *pcard, } if (hw <= OPTi9XX_HW_82C930) chip->mc_base -= 0x80; + + error = snd_opti9xx_read_check(chip); + if (error) { + snd_printk(KERN_ERR "OPTI chip not found\n"); + snd_card_free(card); + return error; + } snd_card_set_dev(card, &pcard->card->dev); if ((error = snd_opti9xx_probe(card)) < 0) { snd_card_free(card); -- 1.6.4 ---------------------------------------------------------------------- Zbieraj punkty - wygrywaj nagrody! Kliknij >>> http://link.interia.pl/f24a8

2 1

[alsa-devel] [PATCH] Use "Skeaper" instead of "chname" by deafult.
by Alexey Fisher 08 Dec '09

08 Dec '09

If you use Speacker by defaut, Surround volume controls will have bad names. For example: before patch -> after patch ----------------------------- Speaker -> Front Speaker 1 -> Surround Speaker 2 -> Side Signed-off-by: Alexey Fisher <bug-track(a)fisher-privat.net> modified: sound/pci/hda/patch_sigmatel.c --- sound/pci/hda/patch_sigmatel.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index e604bae..121a93e 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3152,13 +3152,13 @@ static int create_multi_out_ctls(struct hda_codec *codec, int num_outs, idx = i; break; case AUTO_PIN_SPEAKER_OUT: - name = "Speaker"; - idx = i; - break; - default: name = chname[i]; idx = 0; break; + default: + name = "Speaker"; + idx = i; + break; } err = create_controls_idx(codec, name, idx, nid, 3); if (err < 0) -- 1.6.3.3

2 3

[alsa-devel] line-in on Intel DG45ID
by Alexey Fisher 08 Dec '09

08 Dec '09

Hi, i recently tried to digitalise some tapes and found line-in not enabled on my board (Intel DG45ID). (Codec: IDT 92HD73E1X5) >From the board documentation i know blue jack used for line-in or side surround. Even i use normal stereo i can't enable line-in. So my question, is it really to make a switch between surround and stereo with line-in? I know it should be possible. Or to have this switch for mic input? regards, Alexey ps: cat /proc/asound/card0/codec#2 Codec: IDT 92HD73E1X5 Address: 2 Function Id: 0x1 Vendor Id: 0x111d7676 Subsystem Id: 0x80865002 Revision Id: 0x100202 No Modem Function Group found Default PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Default Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x02, mute=1 GPIO: io=8, o=0, i=0, unsolicited=1, wake=1 IO[0]: enable=1, dir=1, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[4]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[5]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[6]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[7]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Power-Map: 0x65 Analog Loopback: 0x00 Node 0x0a [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x0000173f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x0221401f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=01, enabled=1 Connection: 6 0x15 0x16 0x17 0x18 0x19* 0x1e Node 0x0b [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x0000173f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x02a19020: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=03, enabled=1 Connection: 6 0x15* 0x16 0x17 0x18 0x19 0x1e Node 0x0c [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x00001737: IN OUT Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x01113014: [Jack] Speaker at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0x40: OUT VREF_HIZ Unsolicited: tag=04, enabled=1 Connection: 6 0x15 0x16 0x17 0x18* 0x19 0x1e Node 0x0d [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense Pin Default 0x01114010: [Jack] Speaker at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=05, enabled=1 Connection: 6 0x15* 0x16 0x17 0x18 0x19 0x1e Node 0x0e [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x00001737: IN OUT Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x01a19030: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Connection: 6 0x15* 0x16 0x17 0x18 0x19 0x1e Node 0x0f [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x00000037: IN OUT Detect Trigger ImpSense Pin Default 0x01111012: [Jack] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x40: OUT Unsolicited: tag=06, enabled=1 Connection: 6 0x15 0x16* 0x17 0x18 0x19 0x1e Node 0x10 [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x00000037: IN OUT Detect Trigger ImpSense Pin Default 0x01116011: [Jack] Speaker at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=07, enabled=1 Connection: 6 0x15 0x16 0x17* 0x18 0x19 0x1e Node 0x11 [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x00000037: IN OUT Detect Trigger ImpSense Pin Default 0x40f000f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 6 0x15* 0x16 0x17 0x18 0x19 0x1e Node 0x12 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x00000024: IN Detect Pin Default 0x40f000f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Node 0x13 [Pin Complex] wcaps 0x400003: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x00000020: IN Pin Default 0x40f000f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN Node 0x14 [Pin Complex] wcaps 0x400003: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x00000020: IN Pin Default 0x40f000f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN Node 0x15 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0x69 0x69] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x16 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0xe9 0xe9] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x17 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0x69 0x69] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x18 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x19 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0x69 0x69] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x1a [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x20 Processing caps: benign=0, ncoeff=0 Node 0x1b [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x21 Processing caps: benign=0, ncoeff=0 Node 0x1c [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=1 Amp-Out vals: [0x00] Node 0x1d [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] Connection: 5 0x28 0x29 0x2a 0x2b 0x12 Node 0x1e [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x9f 0x9f] Connection: 1 0x1d Node 0x1f [Volume Knob Widget] wcaps 0x600000: Mono Volume-Knob: delta=1, steps=127, direct=1, val=127 Connection: 5 0x15 0x16 0x17 0x18 0x19 Node 0x20 [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Amp-Out vals: [0x05 0x05] Connection: 12 0x0a 0x0b 0x0c 0x0d 0x0e* 0x0f 0x10 0x11 0x12 0x13 0x14 0x1d Node 0x21 [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 12 0x0a 0x0b 0x0c 0x0d 0x0e* 0x0f 0x10 0x11 0x12 0x13 0x14 0x1d 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: 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 Conn = Optical, Color = Black DefAssociation = 0x4, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 3 0x26* 0x20 0x21 Node 0x24 [Pin Complex] wcaps 0x400681: Stereo Digital Pincap 0x00010024: IN EAPD Detect EAPD 0x0: Pin Default 0x40f000f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Node 0x25 [Audio Output] wcaps 0x4021d: Stereo Digital Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Node 0x26 [Audio Output] wcaps 0x4021d: Stereo Digital Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Node 0x27 [Audio Input] wcaps 0x14031b: Stereo Digital Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Connection: 1 0x24 Node 0x28 [Audio Selector] wcaps 0x300101: Stereo Connection: 4 0x0a* 0x0b 0x0d 0x0f Node 0x29 [Audio Selector] wcaps 0x300101: Stereo Connection: 4 0x0a* 0x0e 0x10 0x11 Node 0x2a [Audio Selector] wcaps 0x300101: Stereo Connection: 4 0x0b* 0x0c 0x10 0x11 Node 0x2b [Audio Selector] wcaps 0x300101: Stereo Connection: 4 0x15* 0x16 0x17 0x18

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[alsa-devel] [PATCH] ASoC: Wrong variable returned on error
by Roel Kluin 08 Dec '09

08 Dec '09

The wrong variable was returned in the case of an error Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- sound/soc/imx/mx1_mx2-pcm.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/imx/mx1_mx2-pcm.c b/sound/soc/imx/mx1_mx2-pcm.c index b838665..bffffcd 100644 --- a/sound/soc/imx/mx1_mx2-pcm.c +++ b/sound/soc/imx/mx1_mx2-pcm.c @@ -322,12 +322,12 @@ static int mx1_mx2_pcm_open(struct snd_pcm_substream *substream) pr_debug("%s: Requesting dma channel (%s)\n", __func__, prtd->dma_params->name); - prtd->dma_ch = imx_dma_request_by_prio(prtd->dma_params->name, - DMA_PRIO_HIGH); - if (prtd->dma_ch < 0) { + ret = imx_dma_request_by_prio(prtd->dma_params->name, DMA_PRIO_HIGH); + if (ret < 0) { printk(KERN_ERR "Error %d requesting dma channel\n", ret); return ret; } + prtd->dma_ch = ret; imx_dma_config_burstlen(prtd->dma_ch, prtd->dma_params->watermark_level);

3 2

[alsa-devel] CS46xx resume patch merge request
by Jaroslav Kysela 08 Dec '09

08 Dec '09

Hi, The new CS46xx resume code is tested now (at least from 2 users): https://bugzilla.redhat.com/show_bug.cgi?id=498287 https://bugzilla.redhat.com/show_bug.cgi?id=160751 Takashi, could you merge topic/cs46xx-resume branch to the development tree? Thanks, Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

2 1

Re: [alsa-devel] io-plugin does not call start callback function
by Stefan Schoenleitner 08 Dec '09

08 Dec '09

Hi, Raul Xiong wrote: > Hi Stefan > > This is because your pointer callback function is not implemented properly. > "pointer" callback function should return the right position of the > playback/capture buffer, apps over alsa framework will call > snd_pcm_avail_update to get the position. > Also, you should call snd_pcm_ioplug_set_param_minmax in your > play_hw_constraint function to define the buffer size relevant parameters. after a full day of experimenting and trying to find out why the start callback function is not called I observed something interesting: It seems that for the io plugin there is an application pointer io->appl_ptr and the hardware pointer io->hw_ptr. Once playback is started, the start callback function is *not* called, but instead data transfer is started immediately. More precisely, each time the pointer callback function is called to get the current position of the hardware pointer and after that the transfer function is called to do the actual data transfer. After each call of the transfer function the number of transfered frames is returned (i.e. the number of frames that have been written to some buffer we would like to fill behind a file descriptor). However, as data transfers have to be timed precisely, we can not just increment the hw_ptr from inside the transfer callback function. If the hw_ptr is not touched at all (thus it stays 0 at all times), a few more transfers are performed until the application pointer appl_ptr has reached a certain (high) amount of "transfered" frames. Then, which is most interesting, the start callback function *is* called followed by calls to the pointer and polling/revent callback functions. As I am not incrementing the hw_ptr at any place right now, the revent polling function is called endlessly. After a full day of even more experiments I think I know how it is supposed to work: In the beginning a certain amount of frames (I guess this is the "start_threshold") is transfered (or more precisely prefilled in the output buffer). As soon as this start threshold is reached the start callback function is called and alsa waits until the hardware pointer hw_ptr has reached a certain position so that the next data transfers can be performed. For this reason it seems that the hw_ptr can be seen as the pointer that indicates how many frames already have been received (and maybe even processed) at the receiving end. Besides it also polls the filedescriptor(s) to check if the next data transfers can be performed (without blocking actually). At the moment I'm not sure if my assumptions above are close enough to reality, but at least I think that I understand what is going on now. If something like this would have been in the alsa documentation I guess it would have saved me like a week of work. cheers, stefan

3 3

Re: [alsa-devel] ALSA full-duplex questions
by Raymond Yau 08 Dec '09

08 Dec '09

2009/11/05 pl bossart <bossart.nospam(a)gmail.com> > > Hi, > I have a very low-*latency* full-duplex application where I would like > up- and downlink to be synchronous, i.e. period events happen at the > same time for capture and playback. This would minimize the number of > wake-ups and enable the platform to go to low power. > > What kind of full-duplex application and how low latency your application need ? > I thought snd_pcm_link() would enable this coupling between capture > and playback, but it looks like it only buids a list of linked devices > and uses this information for state transitions. Said transitions > don't look quite atomic by looking at the code. The > alsa-lib/test/*latency*.c code even tries to measure the difference > between the number of samples read/written. > In short the question is whether I can force a synchronous behavior > from my application, or if I need to roll up my sleeves and fix the > drivers to see this behavior? > > http://thread.gmane.org/gmane.linux.alsa.devel/37367/focus=37388 To get low latency , your application have to use small period size (i.e. your system is unlikely go to low power mode ) BTW. do you know how PA calculate "Final latency" ? D: memblockq.c: memblockq requested: maxlength=4194304, tlength=44100, base=4, prebuf=44100, minreq=3528 maxrewind=0 D: memblockq.c: memblockq sanitized: maxlength=4194304, tlength=44100, base=4, prebuf=44100, minreq=3528 maxrewind=0 I: protocol-native.c: Final latency 460.00 ms = 210.00 ms + 2*20.00 ms + 210.00 ms D: module-alsa-sink.c: latency set to 46.44ms D: module-alsa-sink.c: hwbuf_unused_frames=1 D: module-alsa-sink.c: setting avail_min=4662 D: module-alsa-sink.c: Requesting rewind due to latency change. > Another solution I thought of would be to disable the period events on > one of the devices, and handle both devices at once when one period > event fires. But I am not sure you can disable all events for an audio > device (was on the wish list of Lennart, not sure if it became > reality?) > Thanks for your feedback. > - Pierre > >

1 0

[alsa-devel] [PATCH]PM_QOS-to-use-handle-based-requests-wireless-update 4/5
by mark gross 07 Dec '09

07 Dec '09

Signed-off-by: mark gross <mgross(a)linux.intel.com> --- drivers/net/wireless/ipw2x00/ipw2100.c | 11 ++++++----- net/mac80211/mlme.c | 2 +- 2 files changed, 7 insertions(+), 6 deletions(-) diff --git a/drivers/net/wireless/ipw2x00/ipw2100.c b/drivers/net/wireless/ipw2x00/ipw2100.c index 39d7b7c..0a6a653 100644 --- a/drivers/net/wireless/ipw2x00/ipw2100.c +++ b/drivers/net/wireless/ipw2x00/ipw2100.c @@ -174,6 +174,8 @@ that only one external action is invoked at a time. #define DRV_DESCRIPTION "Intel(R) PRO/Wireless 2100 Network Driver" #define DRV_COPYRIGHT "Copyright(c) 2003-2006 Intel Corporation" +struct pm_qos_request_list *ipw2100_pm_qos_req; + /* Debugging stuff */ #ifdef CONFIG_IPW2100_DEBUG #define IPW2100_RX_DEBUG /* Reception debugging */ @@ -1739,7 +1741,7 @@ static int ipw2100_up(struct ipw2100_priv *priv, int deferred) /* the ipw2100 hardware really doesn't want power management delays * longer than 175usec */ - pm_qos_update_requirement(PM_QOS_CPU_DMA_LATENCY, "ipw2100", 175); + pm_qos_update_request(ipw2100_pm_qos_req, 175); /* If the interrupt is enabled, turn it off... */ spin_lock_irqsave(&priv->low_lock, flags); @@ -1887,8 +1889,7 @@ static void ipw2100_down(struct ipw2100_priv *priv) ipw2100_disable_interrupts(priv); spin_unlock_irqrestore(&priv->low_lock, flags); - pm_qos_update_requirement(PM_QOS_CPU_DMA_LATENCY, "ipw2100", - PM_QOS_DEFAULT_VALUE); + pm_qos_update_request(ipw2100_pm_qos_req, PM_QOS_DEFAULT_VALUE); /* We have to signal any supplicant if we are disassociating */ if (associated) @@ -6658,7 +6659,7 @@ static int __init ipw2100_init(void) if (ret) goto out; - pm_qos_add_requirement(PM_QOS_CPU_DMA_LATENCY, "ipw2100", + ipw2100_pm_qos_req = pm_qos_add_request(PM_QOS_CPU_DMA_LATENCY, PM_QOS_DEFAULT_VALUE); #ifdef CONFIG_IPW2100_DEBUG ipw2100_debug_level = debug; @@ -6681,7 +6682,7 @@ static void __exit ipw2100_exit(void) &driver_attr_debug_level); #endif pci_unregister_driver(&ipw2100_pci_driver); - pm_qos_remove_requirement(PM_QOS_CPU_DMA_LATENCY, "ipw2100"); + pm_qos_remove_request(ipw2100_pm_qos_req); } module_init(ipw2100_init); diff --git a/net/mac80211/mlme.c b/net/mac80211/mlme.c index 6dc7b5a..b8c5caf 100644 --- a/net/mac80211/mlme.c +++ b/net/mac80211/mlme.c @@ -707,7 +707,7 @@ void ieee80211_recalc_ps(struct ieee80211_local *local, s32 latency) s32 beaconint_us; if (latency < 0) - latency = pm_qos_requirement(PM_QOS_NETWORK_LATENCY); + latency = pm_qos_request(PM_QOS_NETWORK_LATENCY); beaconint_us = ieee80211_tu_to_usec( found->vif.bss_conf.beacon_int); -- 1.6.3.3

1 0

[alsa-devel] [PATCH]PM_QOS-to-use-handle-based-requests-network-update 3/5
by mark gross 07 Dec '09

07 Dec '09

Signed-off-by: mark gross <mgross(a)linux.intel.com> --- drivers/net/e1000e/netdev.c | 22 ++++++++++++---------- drivers/net/igbvf/netdev.c | 6 ++++-- include/linux/netdevice.h | 4 ++++ 3 files changed, 20 insertions(+), 12 deletions(-) diff --git a/drivers/net/e1000e/netdev.c b/drivers/net/e1000e/netdev.c index c3105c5..939f8f9 100644 --- a/drivers/net/e1000e/netdev.c +++ b/drivers/net/e1000e/netdev.c @@ -2518,12 +2518,12 @@ static void e1000_configure_rx(struct e1000_adapter *adapter) * excessive C-state transition latencies result in * dropped transactions. */ - pm_qos_update_requirement(PM_QOS_CPU_DMA_LATENCY, - adapter->netdev->name, 55); + pm_qos_update_request( + adapter->netdev->pm_qos_req, 55); } else { - pm_qos_update_requirement(PM_QOS_CPU_DMA_LATENCY, - adapter->netdev->name, - PM_QOS_DEFAULT_VALUE); + pm_qos_update_request( + adapter->netdev->pm_qos_req, + PM_QOS_DEFAULT_VALUE); } } @@ -2833,8 +2833,8 @@ int e1000e_up(struct e1000_adapter *adapter) /* DMA latency requirement to workaround early-receive/jumbo issue */ if (adapter->flags & FLAG_HAS_ERT) - pm_qos_add_requirement(PM_QOS_CPU_DMA_LATENCY, - adapter->netdev->name, + adapter->netdev->pm_qos_req = + pm_qos_add_request(PM_QOS_CPU_DMA_LATENCY, PM_QOS_DEFAULT_VALUE); /* hardware has been reset, we need to reload some things */ @@ -2897,9 +2897,11 @@ void e1000e_down(struct e1000_adapter *adapter) e1000_clean_tx_ring(adapter); e1000_clean_rx_ring(adapter); - if (adapter->flags & FLAG_HAS_ERT) - pm_qos_remove_requirement(PM_QOS_CPU_DMA_LATENCY, - adapter->netdev->name); + if (adapter->flags & FLAG_HAS_ERT) { + pm_qos_remove_request( + adapter->netdev->pm_qos_req); + adapter->netdev->pm_qos_req = NULL; + } /* * TODO: for power management, we could drop the link and diff --git a/drivers/net/igbvf/netdev.c b/drivers/net/igbvf/netdev.c index a127620..6751da8 100644 --- a/drivers/net/igbvf/netdev.c +++ b/drivers/net/igbvf/netdev.c @@ -47,6 +47,7 @@ #define DRV_VERSION "1.0.0-k0" char igbvf_driver_name[] = "igbvf"; const char igbvf_driver_version[] = DRV_VERSION; +struct pm_qos_request_list *igbvf_driver_pm_qos_req; static const char igbvf_driver_string[] = "Intel(R) Virtual Function Network Driver"; static const char igbvf_copyright[] = "Copyright (c) 2009 Intel Corporation."; @@ -2912,7 +2913,7 @@ static int __init igbvf_init_module(void) printk(KERN_INFO "%s\n", igbvf_copyright); ret = pci_register_driver(&igbvf_driver); - pm_qos_add_requirement(PM_QOS_CPU_DMA_LATENCY, igbvf_driver_name, + igbvf_driver_pm_qos_req = pm_qos_add_request(PM_QOS_CPU_DMA_LATENCY, PM_QOS_DEFAULT_VALUE); return ret; @@ -2928,7 +2929,8 @@ module_init(igbvf_init_module); static void __exit igbvf_exit_module(void) { pci_unregister_driver(&igbvf_driver); - pm_qos_remove_requirement(PM_QOS_CPU_DMA_LATENCY, igbvf_driver_name); + pm_qos_remove_request(igbvf_driver_pm_qos_req); + igbvf_driver_pm_qos_req = NULL; } module_exit(igbvf_exit_module); diff --git a/include/linux/netdevice.h b/include/linux/netdevice.h index daf13d3..4437eb5 100644 --- a/include/linux/netdevice.h +++ b/include/linux/netdevice.h @@ -30,6 +30,7 @@ #include <linux/if_packet.h> #ifdef __KERNEL__ +#include <linux/pm_qos_params.h> #include <linux/timer.h> #include <linux/delay.h> #include <linux/mm.h> @@ -693,6 +694,9 @@ struct net_device { * the interface. */ char name[IFNAMSIZ]; + + struct pm_qos_request_list *pm_qos_req; + /* device name hash chain */ struct hlist_node name_hlist; /* snmp alias */ -- 1.6.3.3

1 0

[alsa-devel] [PATCH]PM_QOS-to-use-handle-based-requests-cpuidle-update 2/5
by mark gross 07 Dec '09

07 Dec '09

Signed-off-by: mark gross <mgross(a)linux.intel.com> --- drivers/acpi/processor_idle.c | 2 +- drivers/cpuidle/governors/ladder.c | 2 +- drivers/cpuidle/governors/menu.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/drivers/acpi/processor_idle.c b/drivers/acpi/processor_idle.c index bbd066e..1f524a2 100644 --- a/drivers/acpi/processor_idle.c +++ b/drivers/acpi/processor_idle.c @@ -696,7 +696,7 @@ static int acpi_processor_power_seq_show(struct seq_file *seq, void *offset) "max_cstate: C%d\n" "maximum allowed latency: %d usec\n", pr->power.state ? pr->power.state - pr->power.states : 0, - max_cstate, pm_qos_requirement(PM_QOS_CPU_DMA_LATENCY)); + max_cstate, pm_qos_request(PM_QOS_CPU_DMA_LATENCY)); seq_puts(seq, "states:\n"); diff --git a/drivers/cpuidle/governors/ladder.c b/drivers/cpuidle/governors/ladder.c index a4bec3f..5f93084 100644 --- a/drivers/cpuidle/governors/ladder.c +++ b/drivers/cpuidle/governors/ladder.c @@ -67,7 +67,7 @@ static int ladder_select_state(struct cpuidle_device *dev) struct ladder_device *ldev = &__get_cpu_var(ladder_devices); struct ladder_device_state *last_state; int last_residency, last_idx = ldev->last_state_idx; - int latency_req = pm_qos_requirement(PM_QOS_CPU_DMA_LATENCY); + int latency_req = pm_qos_request(PM_QOS_CPU_DMA_LATENCY); if (unlikely(!ldev)) return 0; diff --git a/drivers/cpuidle/governors/menu.c b/drivers/cpuidle/governors/menu.c index 6810443..f75aec5 100644 --- a/drivers/cpuidle/governors/menu.c +++ b/drivers/cpuidle/governors/menu.c @@ -176,7 +176,7 @@ static void menu_update(struct cpuidle_device *dev); static int menu_select(struct cpuidle_device *dev) { struct menu_device *data = &__get_cpu_var(menu_devices); - int latency_req = pm_qos_requirement(PM_QOS_CPU_DMA_LATENCY); + int latency_req = pm_qos_request(PM_QOS_CPU_DMA_LATENCY); int i; int multiplier; -- 1.6.3.3

1 0

[alsa-devel] [PATCH]PM_QOS-to-use-handle-based-requests-API-change 1/5
by mark gross 07 Dec '09

07 Dec '09

Signed-off-by: mark gross <mgross(a)linux.intel.com> --- include/linux/pm_qos_params.h | 14 ++-- kernel/pm_qos_params.c | 188 ++++++++++++++++++----------------------- 2 files changed, 92 insertions(+), 110 deletions(-) diff --git a/include/linux/pm_qos_params.h b/include/linux/pm_qos_params.h index d74f75e..8ba440e 100644 --- a/include/linux/pm_qos_params.h +++ b/include/linux/pm_qos_params.h @@ -14,12 +14,14 @@ #define PM_QOS_NUM_CLASSES 4 #define PM_QOS_DEFAULT_VALUE -1 -int pm_qos_add_requirement(int qos, char *name, s32 value); -int pm_qos_update_requirement(int qos, char *name, s32 new_value); -void pm_qos_remove_requirement(int qos, char *name); +struct pm_qos_request_list; -int pm_qos_requirement(int qos); +struct pm_qos_request_list *pm_qos_add_request(int pm_qos_class, s32 value); +void pm_qos_update_request(struct pm_qos_request_list *pm_qos_req, + s32 new_value); +void pm_qos_remove_request(struct pm_qos_request_list *pm_qos_req); -int pm_qos_add_notifier(int qos, struct notifier_block *notifier); -int pm_qos_remove_notifier(int qos, struct notifier_block *notifier); +int pm_qos_request(int pm_qos_class); +int pm_qos_add_notifier(int pm_qos_class, struct notifier_block *notifier); +int pm_qos_remove_notifier(int pm_qos_class, struct notifier_block *notifier); diff --git a/kernel/pm_qos_params.c b/kernel/pm_qos_params.c index dfdec52..dc55715 100644 --- a/kernel/pm_qos_params.c +++ b/kernel/pm_qos_params.c @@ -2,7 +2,7 @@ * This module exposes the interface to kernel space for specifying * QoS dependencies. It provides infrastructure for registration of: * - * Dependents on a QoS value : register requirements + * Dependents on a QoS value : register requests * Watchers of QoS value : get notified when target QoS value changes * * This QoS design is best effort based. Dependents register their QoS needs. @@ -14,14 +14,14 @@ * timeout: usec <-- currently not used. * throughput: kbs (kilo byte / sec) * - * There are lists of pm_qos_objects each one wrapping requirements, notifiers + * There are lists of pm_qos_objects each one wrapping requests, notifiers * - * User mode requirements on a QOS parameter register themselves to the + * User mode requests on a QOS parameter register themselves to the * subsystem by opening the device node /dev/... and writing there request to * the node. As long as the process holds a file handle open to the node the * client continues to be accounted for. Upon file release the usermode - * requirement is removed and a new qos target is computed. This way when the - * requirement that the application has is cleaned up when closes the file + * request is removed and a new qos target is computed. This way when the + * request that the application has is cleaned up when closes the file * pointer or exits the pm_qos_object will get an opportunity to clean up. * * Mark Gross <mgross(a)linux.intel.com> @@ -43,25 +43,25 @@ #include <linux/uaccess.h> /* - * locking rule: all changes to requirements or notifiers lists + * locking rule: all changes to requests or notifiers lists * or pm_qos_object list and pm_qos_objects need to happen with pm_qos_lock * held, taken with _irqsave. One lock to rule them all */ -struct requirement_list { +struct pm_qos_request_list { struct list_head list; union { s32 value; s32 usec; s32 kbps; }; - char *name; + int pm_qos_class; }; static s32 max_compare(s32 v1, s32 v2); static s32 min_compare(s32 v1, s32 v2); struct pm_qos_object { - struct requirement_list requirements; + struct pm_qos_request_list requests; struct blocking_notifier_head *notifiers; struct miscdevice pm_qos_power_miscdev; char *name; @@ -73,7 +73,7 @@ struct pm_qos_object { static struct pm_qos_object null_pm_qos; static BLOCKING_NOTIFIER_HEAD(cpu_dma_lat_notifier); static struct pm_qos_object cpu_dma_pm_qos = { - .requirements = {LIST_HEAD_INIT(cpu_dma_pm_qos.requirements.list)}, + .requests = {LIST_HEAD_INIT(cpu_dma_pm_qos.requests.list)}, .notifiers = &cpu_dma_lat_notifier, .name = "cpu_dma_latency", .default_value = 2000 * USEC_PER_SEC, @@ -83,7 +83,7 @@ static struct pm_qos_object cpu_dma_pm_qos = { static BLOCKING_NOTIFIER_HEAD(network_lat_notifier); static struct pm_qos_object network_lat_pm_qos = { - .requirements = {LIST_HEAD_INIT(network_lat_pm_qos.requirements.list)}, + .requests = {LIST_HEAD_INIT(network_lat_pm_qos.requests.list)}, .notifiers = &network_lat_notifier, .name = "network_latency", .default_value = 2000 * USEC_PER_SEC, @@ -94,8 +94,7 @@ static struct pm_qos_object network_lat_pm_qos = { static BLOCKING_NOTIFIER_HEAD(network_throughput_notifier); static struct pm_qos_object network_throughput_pm_qos = { - .requirements = - {LIST_HEAD_INIT(network_throughput_pm_qos.requirements.list)}, + .requests = {LIST_HEAD_INIT(network_throughput_pm_qos.requests.list)}, .notifiers = &network_throughput_notifier, .name = "network_throughput", .default_value = 0, @@ -136,31 +135,34 @@ static s32 min_compare(s32 v1, s32 v2) } -static void update_target(int target) +static void update_target(int pm_qos_class) { s32 extreme_value; - struct requirement_list *node; + struct pm_qos_request_list *node; unsigned long flags; int call_notifier = 0; spin_lock_irqsave(&pm_qos_lock, flags); - extreme_value = pm_qos_array[target]->default_value; + extreme_value = pm_qos_array[pm_qos_class]->default_value; list_for_each_entry(node, - &pm_qos_array[target]->requirements.list, list) { - extreme_value = pm_qos_array[target]->comparitor( + &pm_qos_array[pm_qos_class]->requests.list, list) { + extreme_value = pm_qos_array[pm_qos_class]->comparitor( extreme_value, node->value); } - if (atomic_read(&pm_qos_array[target]->target_value) != extreme_value) { + if (atomic_read(&pm_qos_array[pm_qos_class]->target_value) != + extreme_value) { call_notifier = 1; - atomic_set(&pm_qos_array[target]->target_value, extreme_value); - pr_debug(KERN_ERR "new target for qos %d is %d\n", target, - atomic_read(&pm_qos_array[target]->target_value)); + atomic_set(&pm_qos_array[pm_qos_class]->target_value, + extreme_value); + pr_debug(KERN_ERR "new target for qos %d is %d\n", pm_qos_class, + atomic_read(&pm_qos_array[pm_qos_class]->target_value)); } spin_unlock_irqrestore(&pm_qos_lock, flags); if (call_notifier) - blocking_notifier_call_chain(pm_qos_array[target]->notifiers, - (unsigned long) extreme_value, NULL); + blocking_notifier_call_chain( + pm_qos_array[pm_qos_class]->notifiers, + (unsigned long) extreme_value, NULL); } static int register_pm_qos_misc(struct pm_qos_object *qos) @@ -186,125 +188,110 @@ static int find_pm_qos_object_by_minor(int minor) } /** - * pm_qos_requirement - returns current system wide qos expectation + * pm_qos_request - returns current system wide qos expectation * @pm_qos_class: identification of which qos value is requested * * This function returns the current target value in an atomic manner. */ -int pm_qos_requirement(int pm_qos_class) +int pm_qos_request(int pm_qos_class) { return atomic_read(&pm_qos_array[pm_qos_class]->target_value); } -EXPORT_SYMBOL_GPL(pm_qos_requirement); +EXPORT_SYMBOL_GPL(pm_qos_request); /** - * pm_qos_add_requirement - inserts new qos request into the list + * pm_qos_add_request - inserts new qos request into the list * @pm_qos_class: identifies which list of qos request to us - * @name: identifies the request * @value: defines the qos request * * This function inserts a new entry in the pm_qos_class list of requested qos * performance characteristics. It recomputes the aggregate QoS expectations - * for the pm_qos_class of parameters. + * for the pm_qos_class of parameters, and returns the pm_qos_request list + * element as a handle for use in updating and removal. Call needs to save + * this handle for later use. */ -int pm_qos_add_requirement(int pm_qos_class, char *name, s32 value) +struct pm_qos_request_list *pm_qos_add_request(int pm_qos_class, s32 value) { - struct requirement_list *dep; + struct pm_qos_request_list *dep; unsigned long flags; - dep = kzalloc(sizeof(struct requirement_list), GFP_KERNEL); + dep = kzalloc(sizeof(struct pm_qos_request_list), GFP_KERNEL); if (dep) { if (value == PM_QOS_DEFAULT_VALUE) dep->value = pm_qos_array[pm_qos_class]->default_value; else dep->value = value; - dep->name = kstrdup(name, GFP_KERNEL); - if (!dep->name) - goto cleanup; + dep->pm_qos_class = pm_qos_class; spin_lock_irqsave(&pm_qos_lock, flags); list_add(&dep->list, - &pm_qos_array[pm_qos_class]->requirements.list); + &pm_qos_array[pm_qos_class]->requests.list); spin_unlock_irqrestore(&pm_qos_lock, flags); update_target(pm_qos_class); - - return 0; } -cleanup: - kfree(dep); - return -ENOMEM; + return dep; } -EXPORT_SYMBOL_GPL(pm_qos_add_requirement); +EXPORT_SYMBOL_GPL(pm_qos_add_request); /** - * pm_qos_update_requirement - modifies an existing qos request - * @pm_qos_class: identifies which list of qos request to us - * @name: identifies the request + * pm_qos_update_request - modifies an existing qos request + * @pm_qos_req : handle to list element holding a pm_qos request to use * @value: defines the qos request * - * Updates an existing qos requirement for the pm_qos_class of parameters along + * Updates an existing qos request for the pm_qos_class of parameters along * with updating the target pm_qos_class value. * - * If the named request isn't in the list then no change is made. + * Attempts are made to make this code callable on hot code paths. */ -int pm_qos_update_requirement(int pm_qos_class, char *name, s32 new_value) +void pm_qos_update_request(struct pm_qos_request_list *pm_qos_req, + s32 new_value) { unsigned long flags; - struct requirement_list *node; int pending_update = 0; + s32 temp; spin_lock_irqsave(&pm_qos_lock, flags); - list_for_each_entry(node, - &pm_qos_array[pm_qos_class]->requirements.list, list) { - if (strcmp(node->name, name) == 0) { - if (new_value == PM_QOS_DEFAULT_VALUE) - node->value = - pm_qos_array[pm_qos_class]->default_value; - else - node->value = new_value; - pending_update = 1; - break; - } + if (new_value == PM_QOS_DEFAULT_VALUE) + temp = pm_qos_array[pm_qos_req->pm_qos_class]->default_value; + else + temp = new_value; + + if (temp != pm_qos_req->value) { + pending_update = 1; + pm_qos_req->value = temp; } spin_unlock_irqrestore(&pm_qos_lock, flags); if (pending_update) - update_target(pm_qos_class); - - return 0; + update_target(pm_qos_req->pm_qos_class); } -EXPORT_SYMBOL_GPL(pm_qos_update_requirement); +EXPORT_SYMBOL_GPL(pm_qos_update_request); /** - * pm_qos_remove_requirement - modifies an existing qos request - * @pm_qos_class: identifies which list of qos request to us - * @name: identifies the request + * pm_qos_remove_request - modifies an existing qos request + * @pm_qos_req: handle to request list element * - * Will remove named qos request from pm_qos_class list of parameters and - * recompute the current target value for the pm_qos_class. + * Will remove pm qos request from the list of requests and + * recompute the current target value for the pm_qos_class. Call this + * on slow code paths. */ -void pm_qos_remove_requirement(int pm_qos_class, char *name) +void pm_qos_remove_request(struct pm_qos_request_list *pm_qos_req) { unsigned long flags; - struct requirement_list *node; - int pending_update = 0; + int qos_class; + if (pm_qos_req == NULL) + return; + /* silent return to keep pcm code cleaner */ + + qos_class = pm_qos_req->pm_qos_class; spin_lock_irqsave(&pm_qos_lock, flags); - list_for_each_entry(node, - &pm_qos_array[pm_qos_class]->requirements.list, list) { - if (strcmp(node->name, name) == 0) { - kfree(node->name); - list_del(&node->list); - kfree(node); - pending_update = 1; - break; - } - } + list_del(&pm_qos_req->list); + kfree(pm_qos_req); spin_unlock_irqrestore(&pm_qos_lock, flags); - if (pending_update) - update_target(pm_qos_class); + update_target(qos_class); } -EXPORT_SYMBOL_GPL(pm_qos_remove_requirement); +EXPORT_SYMBOL_GPL(pm_qos_remove_request); /** * pm_qos_add_notifier - sets notification entry for changes to target value @@ -314,7 +301,7 @@ EXPORT_SYMBOL_GPL(pm_qos_remove_requirement); * will register the notifier into a notification chain that gets called * upon changes to the pm_qos_class target value. */ - int pm_qos_add_notifier(int pm_qos_class, struct notifier_block *notifier) +int pm_qos_add_notifier(int pm_qos_class, struct notifier_block *notifier) { int retval; @@ -344,22 +331,17 @@ int pm_qos_remove_notifier(int pm_qos_class, struct notifier_block *notifier) } EXPORT_SYMBOL_GPL(pm_qos_remove_notifier); -#define PID_NAME_LEN sizeof("process_1234567890") -static char name[PID_NAME_LEN]; - static int pm_qos_power_open(struct inode *inode, struct file *filp) { - int ret; long pm_qos_class; lock_kernel(); pm_qos_class = find_pm_qos_object_by_minor(iminor(inode)); + if (pm_qos_class >= 0) { - filp->private_data = (void *)pm_qos_class; - sprintf(name, "process_%d", current->pid); - ret = pm_qos_add_requirement(pm_qos_class, name, - PM_QOS_DEFAULT_VALUE); - if (ret >= 0) { + filp->private_data = (void *) pm_qos_add_request(pm_qos_class, + PM_QOS_DEFAULT_VALUE); + if (filp->private_data) { unlock_kernel(); return 0; } @@ -371,11 +353,10 @@ static int pm_qos_power_open(struct inode *inode, struct file *filp) static int pm_qos_power_release(struct inode *inode, struct file *filp) { - int pm_qos_class; + struct pm_qos_request_list *req; - pm_qos_class = (long)filp->private_data; - sprintf(name, "process_%d", current->pid); - pm_qos_remove_requirement(pm_qos_class, name); + req = (struct pm_qos_request_list *)filp->private_data; + pm_qos_remove_request(req); return 0; } @@ -384,15 +365,14 @@ static ssize_t pm_qos_power_write(struct file *filp, const char __user *buf, size_t count, loff_t *f_pos) { s32 value; - int pm_qos_class; + struct pm_qos_request_list *pm_qos_req; - pm_qos_class = (long)filp->private_data; if (count != sizeof(s32)) return -EINVAL; if (copy_from_user(&value, buf, sizeof(s32))) return -EFAULT; - sprintf(name, "process_%d", current->pid); - pm_qos_update_requirement(pm_qos_class, name, value); + pm_qos_req = (struct pm_qos_request_list *)filp->private_data; + pm_qos_update_request(pm_qos_req, value); return sizeof(s32); } -- 1.6.3.3

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[alsa-devel] [PATCH]PM_QOS-to-use-handle-based-requests 0/5
by mark gross 07 Dec '09

07 Dec '09

This is a 5 patch series against linux-next-20091204 that changes pm_qos to use a handle based access model for the kernel api. The string based model was shown to be slow if pm_qos parameters are updated on hot code paths. Most of the design changes where motivated by a patch aili(a)codeaurora.org sent me, and I spent some time cleaning up the API. Also, I couldn't resist myself in renaming the API's a bit given the fact that the signatures changed enough that I had to touch all the pm_qos users anyway. I changed *requirement* to *request* in keeping with the way PM_QOS really only does best effort. I've felt "requirement" is too strong a word for the way it works. If folks would rather me do the function re-naming in a separate patch set we can do that too. diffstat for the series against linux-next-20091204 : drivers/acpi/processor_idle.c | 2 drivers/cpuidle/governors/ladder.c | 2 drivers/cpuidle/governors/menu.c | 2 drivers/net/e1000e/netdev.c | 22 ++- drivers/net/igbvf/netdev.c | 6 - drivers/net/wireless/ipw2x00/ipw2100.c | 11 + include/linux/netdevice.h | 4 include/linux/pm_qos_params.h | 14 +- include/sound/pcm.h | 3 kernel/pm_qos_params.c | 188 ++++++++++++++------------------- net/mac80211/mlme.c | 2 sound/core/pcm.c | 3 sound/core/pcm_native.c | 14 +- 13 files changed, 132 insertions(+), 141 deletions(-) --mgross

1 0

[alsa-devel] About TASCAM US-122 mkII support
by Tobias Hansen 07 Dec '09

07 Dec '09

Alberto Martinez de la Ossa <delaossa(a)mail.desy.de> wrote: > Hello, > I have just got the TASCAM US-122 mkI > <http://www.tascam.com/products/us-122mkII.html>I audio/midi interface, > which seemed to me a great product and a good deal. > The problem is that I use a linux system (ubuntu-studio karmic) for > music production and the interface can not be recognized when plugged. > Among other reasons I chose this device because I thought it would work > in linux: > There were alsa drivers for its predecessors, the us-122 > <http://alsa.opensrc.org/Tascam_US-122> and the us-122l > <http://www.pps.jussieu.fr/%7Esmimram/tascam/> models. > The latter, thanks to Karsten Wiese who wrote a patch for the us-122l > version, which was latter included by default in newer alsa-firmware > releases. > > Unfortunately, the new model (US-122 mkII), the one I have, is not > supported yet. > I have even tried to get it replaced by the previous us-122l model, but > it is not available anymore in the music store -> http://www.justmusic.de > On the other hand, the "linux specialist" of the music store told me > that a new linux driver for the us-122 mkII is been prepared, > although he doesn't know when it will be ready. > Being you the experts I have found in the web in regards with ALSA and > TASCAM usb interfaces, I have decided to directly ask you about this. > I would really appreciate any information about the topic. Hi, I know that the US-144 MKII has a different Product ID (8020) than the US-144 (800F), so the US-122 MKII probably also has another one than the US-122L. If you post your output of lsusb with the device plugged in, I could make a patch that adds these both IDs to the us122l driver for testing if that suffices. I don't know if you are familiar with compiling kernel modules etc. If you need more detailed instructions, you can join the discussion at http://forum.ubuntuusers.de/topic/tascam-us-144 in either German or English (I know desy.de doesn't necessarily mean you speak German) or send a mail. By the way, I suspect the "linux specialist" at that shop is good in making things up, because Tascam has never provided any Linux drivers and he can't really know what the Linux people do on their own. Best regards, Tobias Hansen

2 1

Re: [alsa-devel] [PATCH] ALSA: ASoc: Add regulator support to CS4270 codec driver
by Mark Brown 07 Dec '09

07 Dec '09

On Sat, Nov 28, 2009 at 07:46:22PM -0700, Tabi Timur-B04825 wrote: > Have you tested it on a Freescale MPC8610 HPCD? Are you sure it still > compiles and builds on any PowerPC system? I can do a compile test on PowerPC (and that's one of the platforms that gets tested with linux-next too) but I don't have hardware to do runtime tests with. Timur, would you be OK with compile tests for now - I'd not expect to see 2.6.33 go out this year so if there are any runtime issues it should be possible to fix them once you're back in the office?

3 11

[alsa-devel] plans to support mixer widget on hda?
by Alexey Fisher 07 Dec '09

07 Dec '09

hallo, are there any plans to support mixer widget? I found it's really nice. Currently i use it with hda-verb for karaoke, to record skype call or just to make some live mix for online radio. Alexey

1 0

[alsa-devel] About TASCAM US-122 mkII support
by Alberto Martinez de la Ossa 07 Dec '09

07 Dec '09

Hello, I have just got the TASCAM US-122 mkI <http://www.tascam.com/products/us-122mkII.html>I audio/midi interface, which seemed to me a great product and a good deal. The problem is that I use a linux system (ubuntu-studio karmic) for music production and the interface can not be recognized when plugged. Among other reasons I chose this device because I thought it would work in linux: There were alsa drivers for its predecessors, the us-122 <http://alsa.opensrc.org/Tascam_US-122> and the us-122l <http://www.pps.jussieu.fr/%7Esmimram/tascam/> models. The latter, thanks to Karsten Wiese who wrote a patch for the us-122l version, which was latter included by default in newer alsa-firmware releases. Unfortunately, the new model (US-122 mkII), the one I have, is not supported yet. I have even tried to get it replaced by the previous us-122l model, but it is not available anymore in the music store -> http://www.justmusic.de On the other hand, the "linux specialist" of the music store told me that a new linux driver for the us-122 mkII is been prepared, although he doesn't know when it will be ready. Being you the experts I have found in the web in regards with ALSA and TASCAM usb interfaces, I have decided to directly ask you about this. I would really appreciate any information about the topic. Thanks in advance for your attention. Best regards,

3 2

Re: [alsa-devel] io-plugin does not call start callback function
by Stefan Schoenleitner 06 Dec '09

06 Dec '09

Hi Raul, in my previous mail I put some snd_pcm_ioplug_set_param_minmax code in the wrong callback function. The corrected code should be below: #include <stdio.h> #include <stdint.h> #include <errno.h> #include <string.h> #include <sys/types.h> #include <sys/stat.h> #include <fcntl.h> #include <pthread.h> #include <time.h> #include <alsa/asoundlib.h> #include <alsa/pcm_external.h> #define FILE_PERM 0644 #define ARRAY_SIZE(a) (sizeof(a)/sizeof((a)[0])) #define DEBUG #ifdef DEBUG #define DBG(f, ...) \ fprintf(stderr, "%s:%i %i "f"\n", __FUNCTION__, __LINE__, getpid(), ## __VA_ARGS__); #else #define DBG(f, ...) #endif struct play_info { snd_pcm_ioplug_t io; char *filename; int file_fd; snd_pcm_sframes_t hw_ptr; }; static int snd_pcm_play_close(snd_pcm_ioplug_t *io) { DBG(""); return 0; } static int snd_pcm_play_start(snd_pcm_ioplug_t *io) { DBG(""); return 0; } static int snd_pcm_play_stop(snd_pcm_ioplug_t *io) { DBG(""); return 0; } static snd_pcm_sframes_t snd_pcm_play_pointer(snd_pcm_ioplug_t *io) { struct play_info *data = io->private_data; DBG("ptr: %lu", data->hw_ptr); return data->hw_ptr; } static snd_pcm_sframes_t snd_pcm_play_transfer(snd_pcm_ioplug_t *io, const snd_pcm_channel_area_t *areas, snd_pcm_uframes_t offset, snd_pcm_uframes_t size) { DBG("aborting with -EINVAL ..."); return -EINVAL; } static int snd_pcm_play_prepare(snd_pcm_ioplug_t *io) { struct play_info *data = io->private_data; DBG("hw_ptr -> 0"); data->hw_ptr=0; return 0; } static int snd_pcm_play_hw_params(snd_pcm_ioplug_t *io, snd_pcm_hw_params_t *params) { return 0; } static int snd_pcm_play_poll_descriptors_count(snd_pcm_ioplug_t *io) { DBG(""); return 1; } static int snd_pcm_play_poll_descriptors(snd_pcm_ioplug_t *io, struct pollfd *pfd, unsigned int space) { DBG(""); return 1; } static int snd_pcm_play_poll_revents(snd_pcm_ioplug_t *io, struct pollfd *pfds, unsigned int nfds, unsigned short *revents) { DBG(""); return 0; } static int snd_pcm_play_delay(snd_pcm_ioplug_t *io, snd_pcm_sframes_t *delayp) { DBG(""); return 0; } static snd_pcm_ioplug_callback_t play_pcm_callback = { .close = snd_pcm_play_close, .start = snd_pcm_play_start, .stop = snd_pcm_play_stop, .pointer = snd_pcm_play_pointer, .transfer = snd_pcm_play_transfer, .prepare = snd_pcm_play_prepare, .hw_params = snd_pcm_play_hw_params, .poll_descriptors_count = snd_pcm_play_poll_descriptors_count, .poll_descriptors = snd_pcm_play_poll_descriptors, .poll_revents = snd_pcm_play_poll_revents, .delay = snd_pcm_play_delay, }; static int play_hw_constraint(struct play_info * pcm) { snd_pcm_ioplug_t *io = &pcm->io; static const snd_pcm_access_t access_list[] = { SND_PCM_ACCESS_RW_INTERLEAVED }; static const unsigned int format_list[] = { SND_PCM_FORMAT_S16_LE, }; int err; DBG(""); /* access type */ err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_ACCESS, ARRAY_SIZE(access_list), access_list); if (err < 0) return err; /* supported formats */ err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_FORMAT, ARRAY_SIZE(format_list), format_list); if (err < 0) return err; /* supported channels */ err = snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_CHANNELS, 1, 1); if (err < 0) return err; /* supported rate */ err = snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_RATE, 8000, 8000); if (err < 0) return err; /* supported block size */ err = snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_PERIOD_BYTES, 156*2, 164*2); if (err < 0) return err; err = snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_PERIODS, 2, 200); if (err < 0) return err; return 0; } SND_PCM_PLUGIN_DEFINE_FUNC(play) { snd_config_iterator_t i, next; const char *filename=NULL; int err; struct play_info *play=NULL; if (stream != SND_PCM_STREAM_PLAYBACK) { SNDERR("play plugin can only be used for playback\n"); return -1; } snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); const char *id; if (snd_config_get_id(n, &id) < 0) continue; if (strcmp(id, "comment") == 0 || strcmp(id, "type") == 0 || strcmp(id, "hint") == 0) continue; if (strcmp(id, "filename") == 0) { if (snd_config_get_type(n) != SND_CONFIG_TYPE_STRING) { SNDERR("invalid type for %s\n", id); return -EINVAL; } if (snd_config_get_string(n, &filename)<0) { SNDERR("could not get filename for %s\n", id); return -EINVAL; } continue; } SNDERR("Unknown field %s", id); return -EINVAL; } if (!filename) { SNDERR("no filename defined or play plugin\n"); return -EINVAL; } if ((play = calloc(1, sizeof(*play))) == NULL) return -ENOMEM; play->io.version = SND_PCM_IOPLUG_VERSION; play->io.name = "ALSA play Plugin"; play->io.callback = &play_pcm_callback; play->io.private_data = play; play->io.mmap_rw = 0; if ((err = snd_pcm_ioplug_create(&play->io, name, stream, mode))<0) { free(play); return err; } err = play_hw_constraint(play); if (err < 0) { snd_pcm_ioplug_delete(&play->io); free(play); return err; } *pcmp = play->io.pcm; return err; } SND_PCM_PLUGIN_SYMBOL(play);

1 0

Re: [alsa-devel] io-plugin does not call start callback function
by Stefan Schoenleitner 06 Dec '09

06 Dec '09

Hi Raul, Raul Xiong wrote: > Hi Stefan > > This is because your pointer callback function is not implemented properly. > "pointer" callback function should return the right position of the > playback/capture buffer, apps over alsa framework will call > snd_pcm_avail_update to get the position. > Also, you should call snd_pcm_ioplug_set_param_minmax in your > play_hw_constraint function to define the buffer size relevant parameters. Well, I did what you suggested but the start() callback is still not called: $ aplay -D play test_8khz_16LE_mono.wav Playing WAVE 'test_8khz_16LE_mono.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono snd_pcm_play_hw_params:84 24610 snd_pcm_play_prepare:71 24610 hw_ptr -> 0 snd_pcm_play_pointer:54 24610 ptr: 0 snd_pcm_play_transfer:63 24610 aborting with -EINVAL ... aplay: pcm_write:1442: write error: Invalid argument Also, when transferring PCM samples from the ALSA buffer to somewhere else, it seems logical to me that the initial value of the hw_ptr has to be 0, right ? As you can see above, the situation is still exactly the same and the start callback is never called at all. The code for the plugin is below. cheers, stefan #include <stdio.h> #include <stdint.h> #include <errno.h> #include <string.h> #include <sys/types.h> #include <sys/stat.h> #include <fcntl.h> #include <pthread.h> #include <time.h> #include <alsa/asoundlib.h> #include <alsa/pcm_external.h> #define FILE_PERM 0644 #define ARRAY_SIZE(a) (sizeof(a)/sizeof((a)[0])) #define DEBUG #ifdef DEBUG #define DBG(f, ...) \ fprintf(stderr, "%s:%i %i "f"\n", __FUNCTION__, __LINE__, getpid(), ## __VA_ARGS__); #else #define DBG(f, ...) #endif struct play_info { snd_pcm_ioplug_t io; char *filename; int file_fd; snd_pcm_sframes_t hw_ptr; snd_pcm_hw_params_t *hw_params; }; static int snd_pcm_play_close(snd_pcm_ioplug_t *io) { DBG(""); return 0; } static int snd_pcm_play_start(snd_pcm_ioplug_t *io) { DBG(""); return 0; } static int snd_pcm_play_stop(snd_pcm_ioplug_t *io) { DBG(""); return 0; } static snd_pcm_sframes_t snd_pcm_play_pointer(snd_pcm_ioplug_t *io) { struct play_info *data = io->private_data; DBG("ptr: %lu", data->hw_ptr); return data->hw_ptr; } static snd_pcm_sframes_t snd_pcm_play_transfer(snd_pcm_ioplug_t *io, const snd_pcm_channel_area_t *areas, snd_pcm_uframes_t offset, snd_pcm_uframes_t size) { DBG("aborting with -EINVAL ..."); return -EINVAL; } static int snd_pcm_play_prepare(snd_pcm_ioplug_t *io) { struct play_info *data = io->private_data; DBG("hw_ptr -> 0"); data->hw_ptr=0; return 0; } static int snd_pcm_play_hw_params(snd_pcm_ioplug_t *io, snd_pcm_hw_params_t *params) { snd_pcm_access_t access_list[] = { SND_PCM_ACCESS_RW_INTERLEAVED }; unsigned int format_list[] = { SND_PCM_FORMAT_S16 }; int err; DBG(""); /* access type */ err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_ACCESS, ARRAY_SIZE(access_list), access_list); if (err < 0) return err; /* supported formats */ err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_FORMAT, ARRAY_SIZE(format_list), format_list); if (err < 0) return err; /* supported channels */ err = snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_CHANNELS, 1, 1); if (err < 0) return err; /* supported rate */ err = snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_RATE, 8000, 8000); if (err < 0) return err; /* supported block size */ err = snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_PERIOD_BYTES, 156*2, 164*2); if (err < 0) return err; err = snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_PERIODS, 2, 200); if (err < 0) return err; return 0; } static int snd_pcm_play_poll_descriptors_count(snd_pcm_ioplug_t *io) { DBG(""); return 1; } static int snd_pcm_play_poll_descriptors(snd_pcm_ioplug_t *io, struct pollfd *pfd, unsigned int space) { DBG(""); return 1; } static int snd_pcm_play_poll_revents(snd_pcm_ioplug_t *io, struct pollfd *pfds, unsigned int nfds, unsigned short *revents) { DBG(""); return 0; } static int snd_pcm_play_delay(snd_pcm_ioplug_t *io, snd_pcm_sframes_t *delayp) { DBG(""); return 0; } static snd_pcm_ioplug_callback_t play_pcm_callback = { .close = snd_pcm_play_close, .start = snd_pcm_play_start, .stop = snd_pcm_play_stop, .pointer = snd_pcm_play_pointer, .transfer = snd_pcm_play_transfer, .prepare = snd_pcm_play_prepare, .hw_params = snd_pcm_play_hw_params, .poll_descriptors_count = snd_pcm_play_poll_descriptors_count, .poll_descriptors = snd_pcm_play_poll_descriptors, .poll_revents = snd_pcm_play_poll_revents, .delay = snd_pcm_play_delay, }; static int play_hw_constraint(struct play_info * pcm) { snd_pcm_ioplug_t *io = &pcm->io; static const snd_pcm_access_t access_list[] = { SND_PCM_ACCESS_RW_INTERLEAVED }; static const unsigned int formats[] = { SND_PCM_FORMAT_S16_LE, }; int err; err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_ACCESS, ARRAY_SIZE(access_list), access_list); if (err < 0) { SNDERR("cannot set HW access constraint list: %s", snd_strerror(err)); return err; } err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_FORMAT, ARRAY_SIZE(formats), formats); if (err < 0) { SNDERR("cannot set HW format constraint list: %s", snd_strerror(err)); return err; } return 0; } SND_PCM_PLUGIN_DEFINE_FUNC(play) { snd_config_iterator_t i, next; const char *filename=NULL; int err; struct play_info *play=NULL; if (stream != SND_PCM_STREAM_PLAYBACK) { SNDERR("play plugin can only be used for playback\n"); return -1; } snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); const char *id; if (snd_config_get_id(n, &id) < 0) continue; if (strcmp(id, "comment") == 0 || strcmp(id, "type") == 0 || strcmp(id, "hint") == 0) continue; if (strcmp(id, "filename") == 0) { if (snd_config_get_type(n) != SND_CONFIG_TYPE_STRING) { SNDERR("invalid type for %s\n", id); return -EINVAL; } if (snd_config_get_string(n, &filename)<0) { SNDERR("could not get filename for %s\n", id); return -EINVAL; } continue; } SNDERR("Unknown field %s", id); return -EINVAL; } if (!filename) { SNDERR("no filename defined or play plugin\n"); return -EINVAL; } if ((play = calloc(1, sizeof(*play))) == NULL) return -ENOMEM; play->io.version = SND_PCM_IOPLUG_VERSION; play->io.name = "ALSA play Plugin"; play->io.callback = &play_pcm_callback; play->io.private_data = play; play->io.mmap_rw = 0; if ((err = snd_pcm_ioplug_create(&play->io, name, stream, mode))<0) { free(play); return err; } err = play_hw_constraint(play); if (err < 0) { snd_pcm_ioplug_delete(&play->io); free(play); return err; } *pcmp = play->io.pcm; return err; } SND_PCM_PLUGIN_SYMBOL(play);

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[alsa-devel] ControlNames: Input Source/ Capture Source/ Jack Mode?
by Alexey Fisher 06 Dec '09

06 Dec '09

Hallo, i have question about controlnames, or if i correctly understand it: Capture Source - you can capture only from one of two (or more) sources. On switching mic a or mic b will be enabled or disabled. /--(o)--mic b cap switch--- \--(x)--mic a Input Source - same as capture source but more verbose /---(Mute/Vol)--Mic a input switch--(a(x)/b(o))-- \---(Mute/Vol)--Mic b more complicated setup: +---(Mute/Vol)--Mic a input switch a-(a(x)/b(o))--+ +---(Mute/Vol)--Mic b input switch b-(a(x)/b(o))--+ jack mode - you can choice to put VREF on the line or not. /---(Mute/Vol)-(mode)-Mic a input switch--(a(x)/b(o))-- \---(Mute/Vol)-(mode)-Mic b i hope i understand it correctly. Now is my situation: +---------+--(Mute/Rec)-(mode)-Mic a input switch a-(a(x)/b(o)/mix(o))--+ | +---------+--(Mute/Rec)-(mode)-Mic b input switch b-(a(x)/b(o)/mix(o))--+ | +--mixer--+<---(Mute/Vol for Mic a/b) are there any way to make alsamixer less verbose but still productive?

1 0

[alsa-devel] ALSA on Alienware M17x
by Adam Gray 06 Dec '09

06 Dec '09

Hi guys, I'm having a lot of trouble getting sound out of my Alienware M17x. I've attached the output from alsa-info. I'm running kernel version 2.6.28 with ALSA driver version 1.0.18. There's no sound at all from the speakers or the headphones. In alsamixer, all channels appear un-muted and turned up full. If you need any further information please let me know. Regards, Adam Gray

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[alsa-devel] when i plugin my headphone the sound come out from speaker and headphone
by Michael Domanski 05 Dec '09

05 Dec '09

Hi, When I plugin my headphone the sound come out from speaker and headphone. The speaker and headphone working in the same time. I want my speaker to mute when headphone is plugged. http://www.alsa-project.org/db/?f=841de9447634e7e35ae0171f968c70de253259af mixer@mdomansky ~ $ sudo lspci |grep -i "multi\|audio\|040[0-3]" 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) mixer@mdomansky ~ $ sudo rmmod snd-hda-intel mixer@mdomansky ~ $ sudo modprobe snd-hda-intel model=hp WARNING: Deprecated config file /etc/modprobe.conf, all config files belong into /etc/modprobe.d/. WARNING: All config files need .conf: /etc/modprobe.d/alsa, it will be ignored in a future release. mixer@mdomansky ~ $ cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0x92400000 irq 22 mixer@mdomansky ~ $ speaker-test -c2 -D hw:0,0 -t wav -l1 ^^^^^ sounds ok ^^^^^^ -- Sincerely, Michael +7 904 512 59 36 MasterWeb.PRO

2 5

[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 04 Dec '09

04 Dec '09

The following changes since commit a649d1fcc9bd2299cb06b6594fabb429fa50f174: Daniel Mack (1): ASoC: pxa/raumfeld: adopt new snd_soc_dai_set_pll() API are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Very last minute but these would qualify as bug fixes during the release cycle so I feel comfortable merging them now. Manuel Lauss (2): ASoC: au1x: dbdma2: fix oops on soc device removal. ASoC: au1x: dbdma2: plug memleak in pcm device creation error path sound/soc/au1x/dbdma2.c | 6 ++---- 1 files changed, 2 insertions(+), 4 deletions(-)

2 3

[alsa-devel] What's in sound git tree for 2.6.33
by Takashi Iwai 04 Dec '09

04 Dec '09

Hi, here is the annual report of sound subsystem for 2.6.33. For 2.6.33, sound bits aren't crazy huge like 2.6.31 or 2.6.32. HD-audio stuff is getting stabilized, but still lots of changes. There are many small fixes here and there, including (not surprisingly) even ISA drivers. Large amounts of changes are found for the support of new VIA HD-audio codecs, Intel HDMI codecs, and a few new drivers and board-supports like Infrasonic Quartet support. In ALSA core side, there are a couple of important fixes. The all "PC Speaker" and "PC Beep" mixer elements are renamed to "Beep". This was a confusion for a long time, and hopefully more understandable now. And, a long-standing PCM mmap problem on PPC and MIPS non-coherent architectures is fixed -- or better to say, Oops is worked around. We are still looking for a saner solution, but it'll be likely an issue for 2.6.34. The biggest change in HD-audio is that we enabled MSI as default now. So far, there is no regression report by this. But, if you see any IRQ issues newly on the new kernel, please report. For ASoC, this has been another fairly quiet release, too. Aside from the addition of virtual mux support to DAPM and some further preparatory work for multi-CODEC cards the majority of changes have been driver updates, including: - New drivers for ADS117x, AK4671, TLV320DAC33, TPA6130A2, WM8711 and WM8727. - Substantial improvements to DMA performance and reliability on MPC5200 and DaVinci. - Capture support for the FSI port on SH. Due to cross tree merge issues some updates to the TWL4030 MFD code are also included, enabling sharing of the CODEC hardware by support for the VIBRA interface of the devices. The tree is at: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus The short log and diffstat are below. Takashi === Akinobu Mita (2): ALSA: ice1712: Use bitrev8 ALSA: sound: usbmidi: Use hweight16 Aleksey Kunitskiy (1): ALSA: ice1724 - Patch for suspend/resume for ESI Juli@ Alexey Fisher (1): ALSA: usb - Fix mixer map for Hercules Gamesurround Muse Pocket LT Anuj Aggarwal (3): ASoC: OMAP3EVM: Use the twl4030_setup_data for headset pop-removal ASoC: Adding OMAP3517 / AM3517 EVM support in ASOC ASoC: Modifying Kconfig/Makefile for AM3517 EVM Barry Song (4): ASoC: Provide API for reordering channels ASoC: use set_channel_map api to reorder channels for AD1938 and AD1836 ASoC: Blackfin I2S: use dai state rather than local counter ASoC: move setting ac97 platformdata earlier than ac97 read/write Ben Dooks (1): ASoC: S3C: Remove <plat/audio.h> Chaithrika U S (1): ASoC: DaVinci: McASP FIFO related updates Clemens Ladisch (23): sound: oxygen: do not try to restore nonexistent EEPROM sound: oxygen: fix for PI7C9X110 compatibility sound: virtuoso: split virtuoso.c sound: virtuoso: fix Xonar Essence ST support sound: oxygen: better defaults for upmixing control sound: oxygen: add stereo upmixing to center/LFE channels sound: oxygen: more hardware documentation sound: oxygen: cache codec registers sound: virtuoso: add headphone impedance control sound: oxygen: allow custom MCLK rates sound: virtuoso: add PCM1796 oversampling control sound: oxygen: add digital filter control sound: oxygen: add high-pass filter control sound: oxygen: fix input monitor control names control: remove snd_konctrol_volatile::owner_pid field control: use reference-counted pid sound: rawmidi: fix opened substreams count sound: pcm: record a substream's owner process sound: rawmidi: record a substream's owner process sound: usb-audio: add Roland UA-1G support sound: usb-audio: allow switching altsetting on Roland USB MIDI devices sound: usb: make the USB MIDI module more independent sound: usxxx: cleanup chip field Dan Carpenter (1): ALSA: remove unnecessary null check Daniel J Blueman (1): mfd: twl4030: fix ELF section mismatch... Daniel Mack (3): ASoC: CS4270: export de-emphasis filter as ALSA control ALSA: ARM: add Raumfeld audio support ASoC: pxa/raumfeld: adopt new snd_soc_dai_set_pll() API Daniel T Chen (2): ALSA: intel8x0: Mute External Amplifier by default for Gateway 4525GZ ALSA: hda: Fix max PCM level to 0 dB for Fujitsu-Siemens laptops using CX20549 (Venice) Eero Nurkkala (2): ASoC: remove io_mutex ASoC: refactor snd_soc_update_bits() Einar Rünkaru (1): ALSA: hda - Make Dell Vostro 1015n mic and speaker switching work Enric Balletbò i Serra (1): ASoC: Add support for IGEP v2 Graeme Gregory (3): ASoC: pxa-ssp increase max_channels to 8 ASoC: ADS117x ADC driver ASoC: omap-mcbsp - add support for upto 16 channels. Grant Likely (6): ASoC/mpc5200: Track DMA position by period number instead of bytes ASoC/mpc5200: get rid of the appl_ptr tracking nonsense ASoC/mpc5200: Improve printk debug output for trigger ASoC/mpc5200: add to_psc_dma_stream() helper ASoC/mpc5200: fix enable/disable of AC97 slots ASoC: mpc5200: remove duplicate identical IRQ handler Igor Grinberg (1): ASoC: finally enable support for eXeda and CM-X300 Janusz Krzysztofik (2): ASoC: Amstrad Delta minor cleanups ASoC: OMAP: Don't try to set unsupported OMAP_DMA_DATA_BURST_16 on OMAP1 Jari Vanhala (1): ASoC: TWL4030: Vibra motor stop fix when it is driven with audio Jarkko Nikula (1): ASoC: Pandora: Pass SRG input clock frequency to the OMAP McBSP DAI Jaroslav Kysela (10): ALSA: rename "PC Speaker" and "PC Beep" controls to "Beep" ALSA: rename "PC Speaker" controls to "Speaker" ALSA: hda - proc - show which I/O NID is associated to PCM device ALSA: hda_intel: Digital PC Beep - change behaviour for input layer ALSA: hda_intel: Digital PC Beep - delay input device unregistration ALSA: hda: beep - add missing cancel_delayed_work ALSA: hda - add beep_mode module parameter ALSA: hda - proc - introduce Control: lines to show mixer<->NID assignment ALSA: hda - proc - add support for dynamic controls to mixer<->NID mapping ALSA: hda - move snd_hda_pcm_type_name from hda_codec.h to hda_local.h Jassi (2): ASoC: S3C I2S LRCLK polarity option. ASoC: Return correct codec clock in s3c64xx-i2s Jassi Brar (7): ASoC: Fixed arguments passed to SMDK64xx set_pll ASoC: Rename s3c24xx_pcm prefix to s3c_dma ASoC: Rename 's3c24xx-pcm' driver to 's3c-dma' ARM: S3C64XX: Define PCM Controller base registers ARM: S3C64XX: Added platform data header ARM: S3C64XX: Defined PCM controller platform devices ASoC: Added the CPU driver for PCM controllers Joonyoung Shim (4): ASoC: Factor out I2C 8 bit address 8 bit data I/O ASoC: AK4671: add ak4671 codec driver ASoC: Add jack_status_check callback function for GPIO jacks ASoC: Rename controls with a / in wm_hubs Julia Lawall (1): ASoC: Move dereference after NULL test Kailang Yang (1): ALSA: hda - Add ALC661/259, ALC892/888VD support Krzysztof Helt (24): ALSA: sscape: add supoort for SPEA Media FX/Reveal SC-600 ALSA: sscape: convert to firmware loader framework ALSA: sscape: remove MIDI instances counting with limit ULONG_MAX ALSA: sscape: force AD1848 codec mode on old Soundscape ALSA: sscape: coding style fixes ALSA: wss: convert CS4231 mixer to dB scale ALSA: wss: reuse CS4231 controls for AD1848 ALSA: snd_dma_pointer workaround for chipsets with buggy DMA sound: remove OSS Ensoniq SoundScape driver ALSA: es18xx: remove snd_card pointer from snd_es18xx structure ALSA: es18xx: remove snd_audiodrive structure ALSA: cs4236: detect chip in one pass ALSA: es18xx: code improvements ALSA: cs4236: update control names ALSA: opti-miro: use variables directly in the probe function ALSA: cs4236: add dB scale for all volume controls ALSA: opti-miro: remove snd_card pointer from snd_miro structure ALSA: opti-miro: make miro.h header available outside the alsa directory ALSA: opti-miro: expose ACI mixer to outside drivers ALSA: opti-miro: fix OOPS if hardware is not detected media/radio: New driver for the radio FM module on Miro PCM20 sound card ALSA: opti-miro: separate comon probing code ALSA: opti-miro: add PnP detection ALSA: opti9xx: remove snd_opti9xx fields Kuninori Morimoto (4): ASoC: sh: FSI: Remove DMA support ASoC: sh: FSI: Add capture support sh: ms7724se: Add runtime PM support for FSI ASoC: sh: fsi: Add runtime PM support Logan Li (1): ALSA: HDA VIA: Remove 48k sample rate limit for S/PDIF Lopez Cruz, Misael (1): ASoC: Add PDM DAI format definition Lydia Wang (30): ALSA: HDA VIA: Remove unused IS_VT17xx_VENDORID macro ALSA: HDA VIA: Change get_codec_type argument to hda_codec type ALSA: HDA VIA: Add VT1708B-CE codec support. ALSA: HDA VIA: Limit VT1702 AA-Path max volume ALSA HDA VIA: Add VIA_CTL_WIDGET_ANALOG_MUTE control type ALSA: HDA VIA: Add low current mode for power saving. ALSA: HDA VIA: Remove unused argument of via_new_analog_input ALSA: HDA VIA: Change VT1708S & VT1702 hp mode controls ALSA: HDA VIA: Rewrite via_independent_hp_put ALSA: HDA VIA: Add smart5.1 function. ALSA: HDA VIA: When changing input source, update power state. ALSA: HDA VIA: Add VIA_JACK_EVENT process in via_unsol_event. ALSA: HDA VIA: Refresh front playback mute in via_hp_automute. ALSA: HDA VIA: Add Jack detect feature for VT1708. ALSA: HDA VIA: Modify vt1708_set_pinconfig_connect function. ALSA: HDA VIA: Replace via_playback_pcm_prepare/cleanup ALSA: HDA VIA: Modify vt1708_auto_create_multi_out_ctls. ALSA: HDA VIA: Modify vt1709_auto_create_multi_out_ctls. ALSA: HDA VIA: Replace MIC_BOOST_VOLUME. ALSA: HDA VIA: Move backdoor verbs to vt17xx_volume_init_verb ALSA: HDA VIA: Add VT1718S support. ALSA: HDA VIA: Add VT1828S and VT2020 support. ALSA: HDA VIA: Add VT1716S support. ALSA: HDA VIA: Add VT2002P support. ALSA: HDA VIA: Add VT1812 support. ALSA: HDA VIA: rename vt1708_control_templates[]. ALSA: HDA VIA: Change PW4 connect select default to to MW0. ALSA: HDA VIA: comments: update copyright, changeset, etc. ALSA: HDA VIA: Only cosmetic changes ALSA: VIA HDA: Add support for VT1818S. Manuel Lauss (5): ASoC: au1x: psc-ac97: verify correct codec register was read ASoC: au1x: psc-ac97: reorganize timeouts ASoC: au1x: convert to platform drivers. ASoC: au1x: dbdma2: fix oops on soc device removal. ASoC: au1x: dbdma2: plug memleak in pcm device creation error path Marin Mitov (1): ASoC: Use DMA_BIT_MASK(32) instead of deprecated DMA_32BIT_MASK Mark Brown (26): ASoC: Update WM8711 to driver model registration method ASoC: Factor out WM8711 cache I/O ASoC: Add SPI support to WM8711 ASoC: WM8711 minor cleanups ASoC: Add TLV information to WM8711 ASoC: Add source argument to PLL configuration ASoC: Allow per-route connectedness checks for supplies ASoC: Add S3C64xx IIS CDCLK source selection ASoC: Convert soc-cache to use C99 style initialisers for the table ASoC: Clean up WM8974 PLL configuration ASoC: Factor out analogue platform data from WM8993 ASoC: Simplify code for DAPM widget updates ASoC: Push DAPM enumeration register change test out ASoC: Add virtual enumeration support for DAPM muxes ASoC: Minor fixups to tpa6130a2 driver ASoC: Remove snd_soc_suspend_device() ASoC: Add regulator support for WM8731 ASoC: Minor SMDK64xx WM8580 cleanups ASoC: Staticise wm8727 driver structure ASoC: Move sysfs and debugfs functions to head of soc-core.c ASoC: Factor out snd_soc_init_card() ASoC: Update ads117x to current APIs ASoC: Add bit clock rate calculator utility functions ASoC: Remove redundant snd_soc_dapm_new_widgets() calls ASoC: Add BCLK calculation utility for TDM mode too S3C64XX: Staticise platform data for PCM devices Miguel Aguilar (1): ASoC: Davinci: Add audio codec support for DM365 EVM Mike Arthur (1): ASoC: Add WM8711 CODEC driver Mike Rapoport (1): ASoC: OMAP: enable Overo driver for CM-T35 Neil Jones (1): ASoC: Add support for the WM8727 DAC. Nicolas Ferre (1): ASoC: at91sam9g20ek_2mmc board uses same audio connexion as at91sam9g20ek Ondrej Zary (2): ALSA: tea575x-tuner: fix mute snd-fm801: autodetect SF64-PCR (tuner-only) card Pavel Hofman (7): ALSA: ak4114 - fix errors in output selector bits ALSA: ak4620 support, codec regs listed in proc ALSA: ak4113 support ALSA: ice1724 - adding GPIO routines for mask and direction ALSA: ice1724 - pro-rate-locking makes sense only for internal clock mode ALSA: ice1724 - Support for multiple external clock types ALSA: ice1724 - Infrasonic Quartet support Peter Ujfalusi (21): ASoC: add support for multiple cards/codecs in debugfs ASoC: Improve the debugfs hierarchy ASoC: TPA6130A2 amplifier driver ASoC: TPA6130A2: Make tpa6130a2_power as static ASoC: Codec driver for Texas Instruments tlv320dac33 codec ASoC: TWL4030: Only update the needed bits in *set_dai_sysclk ASoC: tlv320dac33: typo fix in the header MFD: twl4030: add twl4030_codec MFD as a new child to the core OMAP: Platform support for twl4030_codec MFD ASoC: TWL4030: use the twl4030-codec.h for register descriptions ASoC: TWL4030: Driver registration via twl4030_codec MFD ASoC: TWL4030: Remove bypass tracking ASoC: TWL4030: Change codec_muted to apll_enabled ASoC: TWL4030: Change APLL powering sequence ASoC: TWL4030: Add APLL supply for the capture path ASoC: TWL4030: Make sure, that the codec is powered on startup MFD: TWL4030: Add audio_mclk to the codec platform data OMAP: Configure audio_mclk for twl4030-codec MFD MFD: twl4030-codec: APLL_INFREQ handling in the MFD driver ASoC: TWL4030: Do not modify the APLL_CTL register ASoC: tlv320dac33: Change RT wq to singlethread wq Rafael Ignacio Zurita (1): ALSA: sh: add SuperH DAC audio driver for ALSA V4 Roel Kluin (3): ALSA: Cleanup redundant tests on unsigned sound: OSS: fix error return in dma_ioctl() sound: OSS: keep index within bounds of midi_devs[] Russell King (1): ALSA: AACI cleanup Stas Sergeev (1): ALSA: snd-pcsp: add nopcm mode Stephen Rothwell (1): sound: use semicolons to end statements Takashi Iwai (41): ALSA: hda - Enable MSI as default ASoC: Fix build errors of wm8711.c with SPI ASoC: Fix snd_soc_dai_set_pll() calls in neo1973_*.c ALSA: sscape - Remove sscap_ioctl.h from include/sound/Kbuild ALSA: hda - Clean up name string creation in patch_realtek.c ALSA: sscape - Remove invalid __devinitdata to module parameters ALSA: hda - Add a proper ifdef to a debug code ALSA: hda - Switch to polling mode before disabling MSI ALSA: hda - Reset pins of IDT/STAC codecs at free ASoC: Remove dead code and labels ALSA: hda - Add reboot notifier to each codec ALSA: hda - Add missing export for snd_hda_bus_reboot_notify ALSA: hda - Add power on/off counter ALSA: hda - Don't access invalid substream in proc file ALSA: hda - Fix build error without CONFIG_SND_HDA_HWDEP=y ALSA: hda - Get rid of magic digits for subdev hack ALSA: hda - Add description of beep_mode in ALSA-Configuration.txt ALSA: hda - Update / add kerneldoc comments to exported functions ALSA: hda - Fix beep_mode option value ALSA: hda - Fix build errors with CONFIG_SND_HDA_INPUT_BEEP=n ALSA: hda - Disable default quirk for Sony VAIO with ALC262 codec ALSA: hda - Replace the rest of jack-detections with snd_hda_jack_detect() ALSA: opti-miro: Fix missing semicolon ALSA: hda - Fix detection of dual headphones ALSA: hda - Fix mute-LED sync on HP laptops with IDT92HD83xxx codecs ALSA: hda - Change quirk for Acer Aspire 5930G ALSA: hda - Fix input and jack Kconfig depenencies Revert "ALSA: hda - Change quirk for Acer Aspire 5930G" ALSA: pcm - Use dma_mmap_coherent() if available ALSA: pcm - define snd_pcm_default_page_ops() ALSA: pcm - fix page conversion on non-coherent MIPS arch ALSA: pcm - fix page conversion on non-coherent PPC arch ALSA: Remove old DMA-mmap code from arm/devdma.c ALSA: hda - Don't trigger pin-sense for STAC/IDT codecs ASoC: Revert missing reset_err in wm97*.c ALSA: hda - Add a position_fix quirk for MSI Wind U115 ALSA: hda - Fix Cxt5047 test mode ALSA: hda - Add a pin-fix for FSC Amilo Pi1505 ALSA: hda - Add position_fix quirk for HP dv3 ALSA: aaci - Clean up duplicate code ALSA: hda - Fix memory leaks in the previous patch Timothy Knoll (1): sound: Kconfig typo fix Tobias Hansen (2): ALSA: snd-usb-us122l: add support for US-144 ALSA: snd-usb-us122l: corrent error number for not probing US-144 on ehci-hcd Troy Kisky (5): ASoC: DaVinci: remove requirement that dma_params is 1st in structure ASoC: DaVinci: i2s, reduce underruns by combining into 1 element ASoC: DaVinci: pcm, rename variables in prep for ping/pong ASoC: DaVinci: pcm, fix underrun by using sram ASoC: DaVinci: use edma_pause, edma_resume Wu Fengguang (21): ALSA: hda - select IbexPeak handler for Calpella ALSA: hda - vectorize get_empty_pcm_device() ALSA: hda - allow up to 4 HDMI devices ALSA: hda - convert intelhdmi global references to local parameters ALSA: hda - remove intelhdmi dependency on multiout ALSA: hda - use pcm prepare/cleanup callbacks for intelhdmi ALSA: hda - reorder intelhdmi prepare/cleanup callbacks ALSA: hda - vectorize intelhdmi ALSA: hda - get intelhdmi max channels from widget caps ALSA: hda - auto parse intelhdmi cvt/pin configurations ALSA: hda - remove static intelhdmi configurations ALSA: intelhdmi - fix audio infoframe fill size ALSA: intelhdmi - fix channel mapping slot mask ALSA: intelhdmi - export monitor-presence and ELD-valid status ALSA: hda - introduce snd_hda_jack_detect() and snd_hda_pin_sense() ALSA: intelhdmi - probe for monitor/eld presence at module init time ALSA: intelhdmi - separate out infoframe checksum routine ALSA: intelhdmi - sticky infoframe ALSA: intelhdmi - sticky stream id and format ALSA: intelhdmi - sticky channel count ALSA: hda - show EPSS capability in proc Wu Zhangjin (1): ALSA: SND_CS5535AUDIO: Remove the X86 platform dependency akpm(a)linux-foundation.org (1): sound/pci/hda/patch_via.c: work around gcc-4.0.2 ICE jassi brar (1): ASoC: Support WM8580 based audio subsystem on SMDK64xx machines Łukasz Wojniłowicz (1): ALSA: hda - 4930g mute lfe and side when pluging in headphones --- Documentation/sound/alsa/ALSA-Configuration.txt | 13 +- Documentation/sound/alsa/ControlNames.txt | 3 +- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + arch/arm/mach-davinci/include/mach/asp.h | 8 + arch/arm/mach-omap2/board-3430sdp.c | 10 + arch/arm/mach-omap2/board-omap3beagle.c | 10 + arch/arm/mach-omap2/board-omap3evm.c | 10 + arch/arm/mach-omap2/board-omap3pandora.c | 10 + arch/arm/mach-omap2/board-overo.c | 10 + arch/arm/mach-omap2/board-zoom2.c | 10 + arch/arm/mach-s3c6400/include/mach/map.h | 2 + arch/arm/plat-s3c/include/plat/audio.h | 48 +- arch/arm/plat-s3c/include/plat/devs.h | 3 + arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h | 2 + arch/arm/plat-s3c64xx/dev-audio.c | 101 +- arch/sh/boards/mach-hp6xx/setup.c | 55 + arch/sh/boards/mach-se/7724/setup.c | 3 + arch/sh/include/mach-common/mach/hp6xx.h | 4 + drivers/media/radio/Kconfig | 18 + drivers/media/radio/Makefile | 1 + drivers/media/radio/radio-miropcm20.c | 270 ++ drivers/mfd/Kconfig | 6 + drivers/mfd/Makefile | 1 + drivers/mfd/twl4030-codec.c | 276 ++ drivers/mfd/twl4030-core.c | 16 +- include/linux/i2c/twl4030.h | 19 + include/linux/mfd/twl4030-codec.h | 272 ++ include/sound/Kbuild | 1 - sound/isa/opti9xx/miro.h => include/sound/aci.h | 23 +- include/sound/ak4113.h | 321 ++ include/sound/ak4114.h | 12 +- include/sound/ak4xxx-adda.h | 5 +- include/sound/control.h | 5 +- include/sound/cs4231-regs.h | 1 - include/sound/pcm.h | 3 + include/sound/rawmidi.h | 2 + include/sound/sh_dac_audio.h | 21 + include/sound/soc-dai.h | 14 +- include/sound/soc-dapm.h | 17 + include/sound/soc.h | 15 +- include/sound/sscape_ioctl.h | 21 - include/sound/tlv320dac33-plat.h | 20 + include/sound/tpa6130a2-plat.h | 30 + include/sound/wss.h | 1 - sound/Kconfig | 2 +- sound/arm/Makefile | 2 +- sound/arm/aaci.c | 35 +- sound/arm/devdma.c | 80 - sound/arm/devdma.h | 3 - sound/core/control.c | 9 +- sound/core/isadma.c | 10 +- sound/core/oss/mixer_oss.c | 4 +- sound/core/pcm.c | 6 +- sound/core/pcm_native.c | 83 +- sound/core/rawmidi.c | 17 +- sound/drivers/pcsp/pcsp.c | 32 +- sound/drivers/pcsp/pcsp.h | 2 +- sound/drivers/pcsp/pcsp_mixer.c | 35 +- sound/i2c/cs8427.c | 15 +- sound/i2c/other/Makefile | 3 +- sound/i2c/other/ak4113.c | 639 ++++ sound/i2c/other/ak4xxx-adda.c | 136 +- sound/i2c/other/tea575x-tuner.c | 2 +- sound/isa/Kconfig | 12 +- sound/isa/cmi8330.c | 4 +- sound/isa/cs423x/cs4236.c | 13 +- sound/isa/cs423x/cs4236_lib.c | 241 +- sound/isa/es1688/es1688_lib.c | 2 +- sound/isa/es18xx.c | 221 +- sound/isa/opti9xx/miro.c | 783 +++-- sound/isa/opti9xx/opti92x-ad1848.c | 110 +- sound/isa/sb/sb_mixer.c | 4 +- sound/isa/sscape.c | 727 ++--- sound/isa/wss/wss_lib.c | 105 +- sound/oss/Kconfig | 12 - sound/oss/Makefile | 1 - sound/oss/audio.c | 2 +- sound/oss/midi_synth.c | 2 +- sound/oss/mpu401.c | 2 +- sound/oss/sh_dac_audio.c | 3 - sound/oss/sscape.c | 1480 --------- sound/pci/Kconfig | 1 + sound/pci/ac97/ac97_codec.c | 6 +- sound/pci/ac97/ac97_patch.c | 12 +- sound/pci/azt3328.c | 4 +- sound/pci/ca0106/ca0106_mixer.c | 4 +- sound/pci/ca0106/ca0106_proc.c | 4 +- sound/pci/cmipci.c | 4 +- sound/pci/ctxfi/ctatc.c | 2 +- sound/pci/emu10k1/emu10k1x.c | 3 +- sound/pci/emu10k1/emumixer.c | 4 +- sound/pci/emu10k1/emuproc.c | 4 +- sound/pci/emu10k1/io.c | 2 +- sound/pci/es1938.c | 2 +- sound/pci/fm801.c | 40 +- sound/pci/hda/Kconfig | 13 +- sound/pci/hda/hda_beep.c | 114 +- sound/pci/hda/hda_beep.h | 10 + sound/pci/hda/hda_codec.c | 607 +++- sound/pci/hda/hda_codec.h | 11 + sound/pci/hda/hda_eld.c | 20 +- sound/pci/hda/hda_generic.c | 17 +- sound/pci/hda/hda_hwdep.c | 38 + sound/pci/hda/hda_intel.c | 50 +- sound/pci/hda/hda_local.h | 69 +- sound/pci/hda/hda_proc.c | 70 + sound/pci/hda/patch_analog.c | 61 +- sound/pci/hda/patch_ca0110.c | 4 +- sound/pci/hda/patch_cirrus.c | 31 +- sound/pci/hda/patch_cmedia.c | 4 +- sound/pci/hda/patch_conexant.c | 189 +- sound/pci/hda/patch_intelhdmi.c | 488 +++- sound/pci/hda/patch_realtek.c | 440 ++-- sound/pci/hda/patch_sigmatel.c | 149 +- sound/pci/hda/patch_via.c | 3509 +++++++++++++++++++-- sound/pci/ice1712/Makefile | 2 +- sound/pci/ice1712/ice1712.c | 12 + sound/pci/ice1712/ice1712.h | 14 +- sound/pci/ice1712/ice1724.c | 103 +- sound/pci/ice1712/juli.c | 56 +- sound/pci/ice1712/quartet.c | 1130 +++++++ sound/pci/ice1712/quartet.h | 10 + sound/pci/intel8x0.c | 6 + sound/pci/oxygen/Makefile | 3 +- sound/pci/oxygen/cs2000.h | 83 + sound/pci/oxygen/hifier.c | 61 +- sound/pci/oxygen/oxygen.c | 248 ++- sound/pci/oxygen/oxygen.h | 5 + sound/pci/oxygen/oxygen_lib.c | 29 +- sound/pci/oxygen/oxygen_mixer.c | 52 +- sound/pci/oxygen/oxygen_pcm.c | 19 +- sound/pci/oxygen/virtuoso.c | 1105 +------- sound/pci/oxygen/xonar.h | 50 + sound/pci/oxygen/xonar_cs43xx.c | 434 +++ sound/pci/oxygen/xonar_hdmi.c | 128 + sound/pci/oxygen/xonar_lib.c | 132 + sound/pci/oxygen/xonar_pcm179x.c | 1115 +++++++ sound/ppc/awacs.c | 12 +- sound/ppc/burgundy.c | 8 +- sound/ppc/tumbler.c | 2 +- sound/sh/Kconfig | 8 + sound/sh/Makefile | 2 + sound/sh/sh_dac_audio.c | 453 +++ sound/soc/Makefile | 2 +- sound/soc/atmel/playpaq_wm8510.c | 2 +- sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/au1x/dbdma2.c | 115 +- sound/soc/au1x/psc-ac97.c | 243 +- sound/soc/au1x/psc-i2s.c | 189 +- sound/soc/au1x/psc.h | 7 +- sound/soc/blackfin/bf5xx-ad1836.c | 7 + sound/soc/blackfin/bf5xx-ad1938.c | 9 +- sound/soc/blackfin/bf5xx-i2s.c | 15 +- sound/soc/blackfin/bf5xx-tdm-pcm.c | 9 +- sound/soc/blackfin/bf5xx-tdm.c | 45 +- sound/soc/blackfin/bf5xx-tdm.h | 11 + sound/soc/codecs/Kconfig | 25 + sound/soc/codecs/Makefile | 12 + sound/soc/codecs/ac97.c | 3 - sound/soc/codecs/ad1836.c | 12 - sound/soc/codecs/ad1938.c | 12 - sound/soc/codecs/ad1980.c | 5 - sound/soc/codecs/ad73311.c | 8 - sound/soc/codecs/ads117x.c | 123 + sound/soc/codecs/ads117x.h | 13 + sound/soc/codecs/ak4104.c | 8 - sound/soc/codecs/ak4535.c | 9 - sound/soc/codecs/ak4642.c | 9 - sound/soc/codecs/ak4671.c | 815 +++++ sound/soc/codecs/ak4671.h | 156 + sound/soc/codecs/cs4270.c | 28 +- sound/soc/codecs/cx20442.c | 12 - sound/soc/codecs/pcm3008.c | 9 - sound/soc/codecs/ssm2602.c | 9 - sound/soc/codecs/stac9766.c | 3 - sound/soc/codecs/tlv320aic23.c | 11 +- sound/soc/codecs/tlv320aic26.c | 11 - sound/soc/codecs/tlv320aic3x.c | 11 - sound/soc/codecs/tlv320dac33.c | 1229 ++++++++ sound/soc/codecs/tlv320dac33.h | 267 ++ sound/soc/codecs/tpa6130a2.c | 463 +++ sound/soc/codecs/tpa6130a2.h | 61 + sound/soc/codecs/twl4030.c | 452 ++-- sound/soc/codecs/twl4030.h | 242 +-- sound/soc/codecs/uda134x.c | 9 - sound/soc/codecs/uda1380.c | 9 - sound/soc/codecs/wm8350.c | 32 +- sound/soc/codecs/wm8400.c | 32 +- sound/soc/codecs/wm8510.c | 14 +- sound/soc/codecs/wm8523.c | 26 - sound/soc/codecs/wm8580.c | 30 +- sound/soc/codecs/wm8711.c | 633 ++++ sound/soc/codecs/wm8711.h | 42 + sound/soc/codecs/wm8727.c | 135 + sound/soc/codecs/wm8727.h | 21 + sound/soc/codecs/wm8728.c | 10 - sound/soc/codecs/wm8731.c | 94 +- sound/soc/codecs/wm8750.c | 9 - sound/soc/codecs/wm8753.c | 49 +- sound/soc/codecs/wm8776.c | 43 - sound/soc/codecs/wm8900.c | 34 +- sound/soc/codecs/wm8903.c | 28 - sound/soc/codecs/wm8940.c | 28 +- sound/soc/codecs/wm8960.c | 30 +- sound/soc/codecs/wm8961.c | 27 - sound/soc/codecs/wm8971.c | 11 +- sound/soc/codecs/wm8974.c | 36 +- sound/soc/codecs/wm8988.c | 44 - sound/soc/codecs/wm8990.c | 14 +- sound/soc/codecs/wm8993.c | 49 +- sound/soc/codecs/wm9081.c | 27 - sound/soc/codecs/wm9705.c | 7 - sound/soc/codecs/wm9712.c | 7 - sound/soc/codecs/wm9713.c | 32 +- sound/soc/codecs/wm_hubs.c | 51 +- sound/soc/codecs/wm_hubs.h | 5 + sound/soc/davinci/Kconfig | 4 +- sound/soc/davinci/davinci-evm.c | 7 +- sound/soc/davinci/davinci-i2s.c | 85 +- sound/soc/davinci/davinci-mcasp.c | 18 +- sound/soc/davinci/davinci-mcasp.h | 5 - sound/soc/davinci/davinci-pcm.c | 571 +++- sound/soc/davinci/davinci-pcm.h | 2 + sound/soc/fsl/mpc5200_dma.c | 123 +- sound/soc/fsl/mpc5200_dma.h | 24 +- sound/soc/fsl/mpc5200_psc_ac97.c | 39 +- sound/soc/imx/mx27vis_wm8974.c | 2 +- sound/soc/omap/Kconfig | 23 +- sound/soc/omap/Makefile | 4 + sound/soc/omap/am3517evm.c | 202 ++ sound/soc/omap/ams-delta.c | 4 +- sound/soc/omap/igep0020.c | 148 + sound/soc/omap/omap-mcbsp.c | 63 +- sound/soc/omap/omap3evm.c | 7 + sound/soc/omap/omap3pandora.c | 24 +- sound/soc/omap/overo.c | 4 +- sound/soc/pxa/Kconfig | 12 +- sound/soc/pxa/Makefile | 2 + sound/soc/pxa/magician.c | 2 +- sound/soc/pxa/pxa-ssp.c | 20 +- sound/soc/pxa/raumfeld.c | 335 ++ sound/soc/pxa/zylonite.c | 5 +- sound/soc/s3c24xx/Kconfig | 12 + sound/soc/s3c24xx/Makefile | 6 +- sound/soc/s3c24xx/jive_wm8750.c | 2 +- sound/soc/s3c24xx/ln2440sbc_alc650.c | 2 +- sound/soc/s3c24xx/neo1973_gta02_wm8753.c | 10 +- sound/soc/s3c24xx/neo1973_wm8753.c | 11 +- sound/soc/s3c24xx/{s3c24xx-pcm.c => s3c-dma.c} | 88 +- sound/soc/s3c24xx/{s3c24xx-pcm.h => s3c-dma.h} | 8 +- sound/soc/s3c24xx/s3c-i2s-v2.c | 35 +- sound/soc/s3c24xx/s3c-i2s-v2.h | 4 +- sound/soc/s3c24xx/s3c-pcm.c | 552 ++++ sound/soc/s3c24xx/s3c-pcm.h | 123 + sound/soc/s3c24xx/s3c2412-i2s.c | 7 +- sound/soc/s3c24xx/s3c2443-ac97.c | 13 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 14 +- sound/soc/s3c24xx/s3c24xx_simtec.c | 2 +- sound/soc/s3c24xx/s3c24xx_simtec_hermes.c | 2 +- sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c | 2 +- sound/soc/s3c24xx/s3c24xx_uda134x.c | 2 +- sound/soc/s3c24xx/s3c64xx-i2s.c | 26 +- sound/soc/s3c24xx/s3c64xx-i2s.h | 1 + sound/soc/s3c24xx/smdk2443_wm9710.c | 2 +- sound/soc/s3c24xx/smdk64xx_wm8580.c | 268 ++ sound/soc/s6000/s6000-pcm.c | 4 +- sound/soc/sh/Kconfig | 1 - sound/soc/sh/fsi.c | 271 +- sound/soc/soc-cache.c | 46 +- sound/soc/soc-core.c | 566 ++-- sound/soc/soc-dapm.c | 135 +- sound/soc/soc-jack.c | 6 +- sound/soc/soc-utils.c | 74 + sound/usb/usbaudio.c | 38 +- sound/usb/usbaudio.h | 7 +- sound/usb/usbmidi.c | 208 +- sound/usb/usbmixer_maps.c | 23 + sound/usb/usbquirks.h | 23 + sound/usb/usx2y/us122l.c | 135 +- sound/usb/usx2y/us122l.h | 4 +- sound/usb/usx2y/usX2Yhwdep.c | 8 +- sound/usb/usx2y/usbusx2y.c | 28 +- sound/usb/usx2y/usbusx2y.h | 6 +- sound/usb/usx2y/usbusx2yaudio.c | 34 +- sound/usb/usx2y/usx2yhwdeppcm.c | 8 +- 285 files changed, 21108 insertions(+), 7537 deletions(-) create mode 100644 drivers/media/radio/radio-miropcm20.c create mode 100644 drivers/mfd/twl4030-codec.c create mode 100644 include/linux/mfd/twl4030-codec.h rename sound/isa/opti9xx/miro.h => include/sound/aci.h (85%) create mode 100644 include/sound/ak4113.h create mode 100644 include/sound/sh_dac_audio.h delete mode 100644 include/sound/sscape_ioctl.h create mode 100644 include/sound/tlv320dac33-plat.h create mode 100644 include/sound/tpa6130a2-plat.h delete mode 100644 sound/arm/devdma.c delete mode 100644 sound/arm/devdma.h create mode 100644 sound/i2c/other/ak4113.c delete mode 100644 sound/oss/sscape.c create mode 100644 sound/pci/ice1712/quartet.c create mode 100644 sound/pci/ice1712/quartet.h create mode 100644 sound/pci/oxygen/cs2000.h create mode 100644 sound/pci/oxygen/xonar.h create mode 100644 sound/pci/oxygen/xonar_cs43xx.c create mode 100644 sound/pci/oxygen/xonar_hdmi.c create mode 100644 sound/pci/oxygen/xonar_lib.c create mode 100644 sound/pci/oxygen/xonar_pcm179x.c create mode 100644 sound/sh/sh_dac_audio.c create mode 100644 sound/soc/codecs/ads117x.c create mode 100644 sound/soc/codecs/ads117x.h create mode 100644 sound/soc/codecs/ak4671.c create mode 100644 sound/soc/codecs/ak4671.h create mode 100644 sound/soc/codecs/tlv320dac33.c create mode 100644 sound/soc/codecs/tlv320dac33.h create mode 100644 sound/soc/codecs/tpa6130a2.c create mode 100644 sound/soc/codecs/tpa6130a2.h create mode 100644 sound/soc/codecs/wm8711.c create mode 100644 sound/soc/codecs/wm8711.h create mode 100644 sound/soc/codecs/wm8727.c create mode 100644 sound/soc/codecs/wm8727.h create mode 100644 sound/soc/omap/am3517evm.c create mode 100644 sound/soc/omap/igep0020.c create mode 100644 sound/soc/pxa/raumfeld.c rename sound/soc/s3c24xx/{s3c24xx-pcm.c => s3c-dma.c} (82%) rename sound/soc/s3c24xx/{s3c24xx-pcm.h => s3c-dma.h} (87%) create mode 100644 sound/soc/s3c24xx/s3c-pcm.c create mode 100644 sound/soc/s3c24xx/s3c-pcm.h create mode 100644 sound/soc/s3c24xx/smdk64xx_wm8580.c create mode 100644 sound/soc/soc-utils.c

1 0

[alsa-devel] Underruns in basic audio playback application
by Devin Heitmueller 04 Dec '09

04 Dec '09

Hello all, I am working on a very basic application that reads a capture device and outputs to a playback device. It's what you would traditionally accomplish with something like the following: arecord -D hw:1,0 -r 48000 -c 2 -f S16_LE | aplay - However, in this case, since the final result will be part of a TV watching application, I need to maintain less than 30ms of latency to preserve lipsync. The application I ended up with essentially creates two PCM streams (one for capture, one for playback), sets them up with the same parameters in terms of rate, format, channels, etc., and then has a loop of snd_pcm_readi() and snd_pcm_writei() calls. The application works, except I am getting a considerable amount of underruns on the playback device. Given my relative inexperience with alsa-lib, I suspect that I have misconfigured one of the parameters effecting buffering - buffer size, buffer time, period size, period time. Can anyone offer any constructive suggestions or tips on what these tuning parameters should be set to for a continuous 48000Khz, 2 channel stream of audio? Also, should the buffering configuration really be the same for both the capture and playback stream, or do I need to play some games such that one requires more/less buffering than the other? Thanks in advance, Devin -- Devin J. Heitmueller - Kernel Labs http://www.kernellabs.com

2 2

[alsa-devel] [PATCH] ASoC: tlv320dac33: Add support for regulator framework
by Ilkka Koskinen 04 Dec '09

04 Dec '09

Take the regulator framework in use for managing the power sources. Signed-off-by: Ilkka Koskinen <ilkka.koskinen(a)nokia.com> --- sound/soc/codecs/tlv320dac33.c | 92 ++++++++++++++++++++++++++++++++++------ 1 files changed, 79 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 9c8903d..5037454 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -30,6 +30,7 @@ #include <linux/platform_device.h> #include <linux/interrupt.h> #include <linux/gpio.h> +#include <linux/regulator/consumer.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -58,11 +59,19 @@ enum dac33_state { DAC33_FLUSH, }; +#define DAC33_NUM_SUPPLIES 3 +static const char *dac33_supply_names[DAC33_NUM_SUPPLIES] = { + "AVDD", + "DVDD", + "IOVDD", +}; + struct tlv320dac33_priv { struct mutex mutex; struct workqueue_struct *dac33_wq; struct work_struct work; struct snd_soc_codec codec; + struct regulator_bulk_data supplies[DAC33_NUM_SUPPLIES]; int power_gpio; int chip_power; int irq; @@ -297,28 +306,49 @@ static inline void dac33_soft_power(struct snd_soc_codec *codec, int power) dac33_write(codec, DAC33_PWR_CTRL, reg); } -static void dac33_hard_power(struct snd_soc_codec *codec, int power) +static int dac33_hard_power(struct snd_soc_codec *codec, int power) { struct tlv320dac33_priv *dac33 = codec->private_data; + int ret; mutex_lock(&dac33->mutex); if (power) { - if (dac33->power_gpio >= 0) { - gpio_set_value(dac33->power_gpio, 1); - dac33->chip_power = 1; - /* Restore registers */ - dac33_restore_regs(codec); + ret = regulator_bulk_enable(ARRAY_SIZE(dac33->supplies), + dac33->supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", ret); + goto exit; } + + if (dac33->power_gpio >= 0) + gpio_set_value(dac33->power_gpio, 1); + + dac33->chip_power = 1; + + /* Restore registers */ + dac33_restore_regs(codec); + dac33_soft_power(codec, 1); } else { dac33_soft_power(codec, 0); - if (dac33->power_gpio >= 0) { + if (dac33->power_gpio >= 0) gpio_set_value(dac33->power_gpio, 0); - dac33->chip_power = 0; + + ret = regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), + dac33->supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to disable supplies: %d\n", ret); + goto exit; } + + dac33->chip_power = 0; } - mutex_unlock(&dac33->mutex); +exit: + mutex_unlock(&dac33->mutex); + return ret; } static int dac33_get_nsample(struct snd_kcontrol *kcontrol, @@ -469,6 +499,8 @@ static int dac33_add_widgets(struct snd_soc_codec *codec) static int dac33_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + int ret; + switch (level) { case SND_SOC_BIAS_ON: dac33_soft_power(codec, 1); @@ -476,12 +508,19 @@ static int dac33_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) - dac33_hard_power(codec, 1); + if (codec->bias_level == SND_SOC_BIAS_OFF) { + ret = dac33_hard_power(codec, 1); + if (ret != 0) + return ret; + } + dac33_soft_power(codec, 0); break; case SND_SOC_BIAS_OFF: - dac33_hard_power(codec, 0); + ret = dac33_hard_power(codec, 0); + if (ret != 0) + return ret; + break; } codec->bias_level = level; @@ -959,6 +998,9 @@ static int dac33_soc_probe(struct platform_device *pdev) /* power on device */ dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* Bias level configuration has enabled regulator an extra time */ + regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), dac33->supplies); + return 0; pcm_err: @@ -1039,7 +1081,7 @@ static int dac33_i2c_probe(struct i2c_client *client, struct tlv320dac33_platform_data *pdata; struct tlv320dac33_priv *dac33; struct snd_soc_codec *codec; - int ret = 0; + int ret, i; if (client->dev.platform_data == NULL) { dev_err(&client->dev, "Platform data not set\n"); @@ -1130,6 +1172,24 @@ static int dac33_i2c_probe(struct i2c_client *client, } } + for (i = 0; i < ARRAY_SIZE(dac33->supplies); i++) + dac33->supplies[i].supply = dac33_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(dac33->supplies), + dac33->supplies); + + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err_get; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(dac33->supplies), + dac33->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_enable; + } + ret = snd_soc_register_codec(codec); if (ret != 0) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); @@ -1149,6 +1209,10 @@ static int dac33_i2c_probe(struct i2c_client *client, return ret; error_codec: + regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), dac33->supplies); +err_enable: + regulator_bulk_free(ARRAY_SIZE(dac33->supplies), dac33->supplies); +err_get: if (dac33->irq >= 0) { free_irq(dac33->irq, &dac33->codec); destroy_workqueue(dac33->dac33_wq); @@ -1177,6 +1241,8 @@ static int dac33_i2c_remove(struct i2c_client *client) if (dac33->irq >= 0) free_irq(dac33->irq, &dac33->codec); + regulator_bulk_free(ARRAY_SIZE(dac33->supplies), dac33->supplies); + destroy_workqueue(dac33->dac33_wq); snd_soc_unregister_dai(&dac33_dai); snd_soc_unregister_codec(&dac33->codec); -- 1.6.0.4

3 2

[alsa-devel] ALSA Control Questions
by Tobias Schneider 04 Dec '09

04 Dec '09

I've got some questions concerning the ALSA control interface... 1. What is a "simple" control? I would like to use the control interface to set more than just volume (e.g. I would like to set input gain, output attenuation, input channel selection, output channel selection, output channel activation...), so I am wondering if I should use SNDRV_CTL_ELEM_IFACE_CARD or SNDRV_CTL_ELEM_IFACE_MIXER, as they are obviously in context with "simple" control... 2. The more interesting question: as written in "Writing an alsa driver" callbacks are basically not atomic, but I am getting BUG: scheduling while atomic: amixer/0x00000001/430 Call Trace: .[..] in "put" callback?? So what is meant with "basically not atomic"?? What is allowed and what is not allowed, am I able to get hardware interrupts in this context?

3 7

[alsa-devel] [PATCH] ASoC: tlv320dac33: Add support for regulator framework
by Ilkka Koskinen 04 Dec '09

04 Dec '09

Take the regulator framework in use for managing the power sources. Signed-off-by: Ilkka Koskinen <ilkka.koskinen(a)nokia.com> --- sound/soc/codecs/tlv320dac33.c | 92 ++++++++++++++++++++++++++++++++++------ 1 files changed, 79 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 3ca8934..c2510bb 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -30,6 +30,7 @@ #include <linux/platform_device.h> #include <linux/interrupt.h> #include <linux/gpio.h> +#include <linux/regulator/consumer.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -58,11 +59,19 @@ enum dac33_state { DAC33_FLUSH, }; +#define DAC33_NUM_SUPPLIES 3 +static const char *dac33_supply_names[DAC33_NUM_SUPPLIES] = { + "AVDD", + "DVDD", + "IOVDD", +}; + struct tlv320dac33_priv { struct mutex mutex; struct workqueue_struct *dac33_wq; struct work_struct work; struct snd_soc_codec codec; + struct regulator_bulk_data supplies[DAC33_NUM_SUPPLIES]; int power_gpio; int chip_power; int irq; @@ -297,28 +306,49 @@ static inline void dac33_soft_power(struct snd_soc_codec *codec, int power) dac33_write(codec, DAC33_PWR_CTRL, reg); } -static void dac33_hard_power(struct snd_soc_codec *codec, int power) +static int dac33_hard_power(struct snd_soc_codec *codec, int power) { struct tlv320dac33_priv *dac33 = codec->private_data; + int ret; mutex_lock(&dac33->mutex); if (power) { - if (dac33->power_gpio >= 0) { - gpio_set_value(dac33->power_gpio, 1); - dac33->chip_power = 1; - /* Restore registers */ - dac33_restore_regs(codec); + ret = regulator_bulk_enable(ARRAY_SIZE(dac33->supplies), + dac33->supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", ret); + goto exit; } + + if (dac33->power_gpio >= 0) + gpio_set_value(dac33->power_gpio, 1); + + dac33->chip_power = 1; + + /* Restore registers */ + dac33_restore_regs(codec); + dac33_soft_power(codec, 1); } else { dac33_soft_power(codec, 0); - if (dac33->power_gpio >= 0) { + if (dac33->power_gpio >= 0) gpio_set_value(dac33->power_gpio, 0); - dac33->chip_power = 0; + + ret = regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), + dac33->supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to disable supplies: %d\n", ret); + goto exit; } + + dac33->chip_power = 0; } - mutex_unlock(&dac33->mutex); +exit: + mutex_unlock(&dac33->mutex); + return ret; } static int dac33_get_nsample(struct snd_kcontrol *kcontrol, @@ -470,6 +500,8 @@ static int dac33_add_widgets(struct snd_soc_codec *codec) static int dac33_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + int ret; + switch (level) { case SND_SOC_BIAS_ON: dac33_soft_power(codec, 1); @@ -477,12 +509,19 @@ static int dac33_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) - dac33_hard_power(codec, 1); + if (codec->bias_level == SND_SOC_BIAS_OFF) { + ret = dac33_hard_power(codec, 1); + if (ret != 0) + return ret; + } + dac33_soft_power(codec, 0); break; case SND_SOC_BIAS_OFF: - dac33_hard_power(codec, 0); + ret = dac33_hard_power(codec, 0); + if (ret != 0) + return ret; + break; } codec->bias_level = level; @@ -960,6 +999,9 @@ static int dac33_soc_probe(struct platform_device *pdev) /* power on device */ dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* Bias level configuration has enabled regulator an extra time */ + regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), dac33->supplies); + ret = snd_soc_init_card(socdev); if (ret < 0) { dev_err(codec->dev, "failed to register card\n"); @@ -1048,7 +1090,7 @@ static int dac33_i2c_probe(struct i2c_client *client, struct tlv320dac33_platform_data *pdata; struct tlv320dac33_priv *dac33; struct snd_soc_codec *codec; - int ret = 0; + int ret, i; if (client->dev.platform_data == NULL) { dev_err(&client->dev, "Platform data not set\n"); @@ -1138,6 +1180,24 @@ static int dac33_i2c_probe(struct i2c_client *client, } } + for (i = 0; i < ARRAY_SIZE(dac33->supplies); i++) + dac33->supplies[i].supply = dac33_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(dac33->supplies), + dac33->supplies); + + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err_get; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(dac33->supplies), + dac33->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_enable; + } + ret = snd_soc_register_codec(codec); if (ret != 0) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); @@ -1157,6 +1217,10 @@ static int dac33_i2c_probe(struct i2c_client *client, return ret; error_codec: + regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), dac33->supplies); +err_enable: + regulator_bulk_free(ARRAY_SIZE(dac33->supplies), dac33->supplies); +err_get: if (dac33->irq >= 0) { free_irq(dac33->irq, &dac33->codec); destroy_workqueue(dac33->dac33_wq); @@ -1185,6 +1249,8 @@ static int dac33_i2c_remove(struct i2c_client *client) if (dac33->irq >= 0) free_irq(dac33->irq, &dac33->codec); + regulator_bulk_free(ARRAY_SIZE(dac33->supplies), dac33->supplies); + destroy_workqueue(dac33->dac33_wq); snd_soc_unregister_dai(&dac33_dai); snd_soc_unregister_codec(&dac33->codec); -- 1.6.0.4

3 2

[alsa-devel] [PATCH v2] ASoC: DaVinci: Update suspend/resume support for McASP driver
by Chaithrika U S 04 Dec '09

04 Dec '09

Add clock enable and disable calls to resume and suspend respectively. Also add a member to the audio device data structure which tracks the clock status. Tested on DA850/OMAP-L138 EVM. For the purpose of testing, the patches[1] which add suspend-to-RAM support to DA850/OMAP-L138 SoC were applied. [1] http://linux.davincidsp.com/pipermail/davinci-linux-open-source/ 2009-November/016958.html Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies to ALSA GIT tree on branch topic/asoc at http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=shortlog; h=topic/asoc sound/soc/davinci/davinci-mcasp.c | 18 ++++++++++++++++-- sound/soc/davinci/davinci-mcasp.h | 1 + sound/soc/davinci/davinci-pcm.c | 2 +- 3 files changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 0a302e1..a613bbb 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -767,14 +767,27 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, int ret = 0; switch (cmd) { - case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: + if (!dev->clk_active) { + clk_enable(dev->clk); + dev->clk_active = 1; + } + /* Fall through */ + case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: davinci_mcasp_start(dev, substream->stream); break; - case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: + davinci_mcasp_stop(dev, substream->stream); + if (dev->clk_active) { + clk_disable(dev->clk); + dev->clk_active = 0; + } + + break; + + case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: davinci_mcasp_stop(dev, substream->stream); break; @@ -866,6 +879,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) } clk_enable(dev->clk); + dev->clk_active = 1; dev->base = (void __iomem *)IO_ADDRESS(mem->start); dev->op_mode = pdata->op_mode; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 582c924..e755b51 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -44,6 +44,7 @@ struct davinci_audio_dev { int sample_rate; struct clk *clk; unsigned int codec_fmt; + u8 clk_active; /* McASP specific data */ int tdm_slots; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index ad4d7f4..80c7fdf 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -49,7 +49,7 @@ static void print_buf_info(int slot, char *name) static struct snd_pcm_hardware pcm_hardware_playback = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE), + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), .formats = (SNDRV_PCM_FMTBIT_S16_LE), .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | -- 1.5.6

2 1

[alsa-devel] snd_hda_intel: MacBook Pro 5,1 / ALC889A issues
by Alexey Spiridonov 04 Dec '09

04 Dec '09

Hello, The hda_intel driver is only marginally usable on my laptop. I described the problems in detail here: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/488103 In brief, one has to choose between two sets of problems: 1) model=mbp3: broken speakers or 2) model=mb5: working speakers, but noisy headphone output Neither option provides headphone jack sense. Neither option is able to access the headset-type microphone available under MacOS through 3-band jacks like this: http://img.tomshardware.com/us/2007/07/19/how_good_is_apples_wired_iphone_h… (MacOS automatically switches from the in-computer mic to the the headset mic -- I haven't been able to get any signal from it despite trying all permutations of alsamixer's capture settings) I have attached the output of alsa-info.sh. I'd be happy to help test patches, or other debugging support. Thanks a lot! ~a

1 0

[alsa-devel] syslog says snd-usb-audio bug?
by Achilles H 04 Dec '09

04 Dec '09

Dec 4 00:16:21 susbwoy pulseaudio[1719]: alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! Dec 4 00:16:21 susbwoy pulseaudio[1719]: alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Dec 4 00:16:21 susbwoy pulseaudio[1719]: alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. I am using Ubuntu 9.10 64bit with an Edirol UA-25 USB sound card. I installed OpenArena, and tried to open it. Screen went black - Could only get back to the desktop by killing X. This is what happened in the terminal when OpenArena Died at Initializing sound ------ Initializing Sound ------ bt_audio_service_open: connect() failed: Connection refused (111) bt_audio_service_open: connect() failed: Connection refused (111) bt_audio_service_open: connect() failed: Connection refused (111) bt_audio_service_open: connect() failed: Connection refused (111) Allocated 96 sources. OpenAL default capture device is 'ALSA Capture on default' AL lib: alsa.c:675: set access failed: Invalid argument AL lib: oss.c:307: Could not open /dev/dsp: No such file or directory Killed I have no issues with sound at all apart from trying to use OpenAL.Warsow(another great game) will also refuse to run with OpenAL, except it runs great in the default sound mode 'qf'. What is going on here? thanks

1 0

[alsa-devel] [PATCH] ASoC: Rename controls with a / in wm_hubs
by Joonyoung Shim 04 Dec '09

04 Dec '09

This renames from a character / to : of controls. A / occurs below error messages. ASoC: Failed to create IN2RP/VXRP debugfs file ASoC: Failed to create IN2LP/VXRN debugfs file Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- If there is the better a delimiter or other idea, please inform me it. sound/soc/codecs/wm_hubs.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 810a563..d73c305 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -438,11 +438,11 @@ static const struct snd_soc_dapm_widget analogue_dapm_widgets[] = { SND_SOC_DAPM_INPUT("IN1LN"), SND_SOC_DAPM_INPUT("IN1LP"), SND_SOC_DAPM_INPUT("IN2LN"), -SND_SOC_DAPM_INPUT("IN2LP/VXRN"), +SND_SOC_DAPM_INPUT("IN2LP:VXRN"), SND_SOC_DAPM_INPUT("IN1RN"), SND_SOC_DAPM_INPUT("IN1RP"), SND_SOC_DAPM_INPUT("IN2RN"), -SND_SOC_DAPM_INPUT("IN2RP/VXRP"), +SND_SOC_DAPM_INPUT("IN2RP:VXRP"), SND_SOC_DAPM_MICBIAS("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0), SND_SOC_DAPM_MICBIAS("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0), @@ -537,14 +537,14 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "IN1R PGA", "IN1RP Switch", "IN1RP" }, { "IN1R PGA", "IN1RN Switch", "IN1RN" }, - { "IN2L PGA", "IN2LP Switch", "IN2LP/VXRN" }, + { "IN2L PGA", "IN2LP Switch", "IN2LP:VXRN" }, { "IN2L PGA", "IN2LN Switch", "IN2LN" }, - { "IN2R PGA", "IN2RP Switch", "IN2RP/VXRP" }, + { "IN2R PGA", "IN2RP Switch", "IN2RP:VXRP" }, { "IN2R PGA", "IN2RN Switch", "IN2RN" }, - { "Direct Voice", NULL, "IN2LP/VXRN" }, - { "Direct Voice", NULL, "IN2RP/VXRP" }, + { "Direct Voice", NULL, "IN2LP:VXRN" }, + { "Direct Voice", NULL, "IN2RP:VXRP" }, { "MIXINL", "IN1L Switch", "IN1L PGA" }, { "MIXINL", "IN2L Switch", "IN2L PGA" }, @@ -565,7 +565,7 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "Left Output Mixer", "Right Input Switch", "MIXINR" }, { "Left Output Mixer", "IN2RN Switch", "IN2RN" }, { "Left Output Mixer", "IN2LN Switch", "IN2LN" }, - { "Left Output Mixer", "IN2LP Switch", "IN2LP/VXRN" }, + { "Left Output Mixer", "IN2LP Switch", "IN2LP:VXRN" }, { "Left Output Mixer", "IN1L Switch", "IN1L PGA" }, { "Left Output Mixer", "IN1R Switch", "IN1R PGA" }, @@ -573,7 +573,7 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "Right Output Mixer", "Right Input Switch", "MIXINR" }, { "Right Output Mixer", "IN2LN Switch", "IN2LN" }, { "Right Output Mixer", "IN2RN Switch", "IN2RN" }, - { "Right Output Mixer", "IN2RP Switch", "IN2RP/VXRP" }, + { "Right Output Mixer", "IN2RP Switch", "IN2RP:VXRP" }, { "Right Output Mixer", "IN1L Switch", "IN1L PGA" }, { "Right Output Mixer", "IN1R Switch", "IN1R PGA" }, -- 1.6.3.3

1 0

[alsa-devel] 48kHz SPDIF out on X-Fi?
by Lucas Marshall 04 Dec '09

04 Dec '09

Is it possible to get the optical SPDIF on an X-Fi Titanium to output at 48kHz instead of 96kHz? I've tried doing a rate convert in .asoundrc, adding a rate to the hw plug in there, etc. but no matter what it's still spewing 96kHz according to my test receiver. We would like to do a SPDIF-to-AES conversion, and our video/audio router won't take anything but 48kHz. :( Thanks in advance for any suggestions! -- Lucas Marshall Systems Administrator // Media Systems Lucasfilm 415-746-2861

1 0

Re: [alsa-devel] [Alsa-user] m-audio delta 1010 inputs crackle at 96 kHz sample rate with all newer kernels
by francis keyes 03 Dec '09

03 Dec '09

The first kernel that broke was a newer 64 bit one. I switched to 32 bit Linux a few months later to try it out because I had other problems. However, this particular problem persists. On Wed, Dec 2, 2009 at 3:49 AM, Clemens Ladisch <clemens(a)alsa-project.org>wrote: > francis keyes wrote: > > The last kernel that worked properly for me was > > > > 2.6.24.7-1.rt21.1.fc9.ccrma.x86_64.rt > > > > (I am still using the same hardware but I no longer use 64 bit linux) > > So the first kernel that broke was the same version in 32 bits, or some > newer one? > > > Best regards, > Clemens >

1 0

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

03 Dec '09

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

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

03 Dec '09

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

2 1

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

03 Dec '09

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

1 0

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

03 Dec '09

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

1 1

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

03 Dec '09

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

4 6

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

03 Dec '09

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

1 0

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

03 Dec '09

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

2 2

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

03 Dec '09

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

3 5

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

03 Dec '09

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

2 3

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

03 Dec '09

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

2 1

[alsa-devel] testing mic-in for tlv320aic33
by vivek kaushik 03 Dec '09

03 Dec '09

Hi All, I am testing MIC IN on tlv320aic33. but getting the following error: arecord: pcm_read:1338: read error: Input/output error On my custom board tlv320aic33 is connected mcbsp-1 of omap3530, i am configuring tlv as mater and omap3530 as slave.aplay is working fine for me but getting the above error in case of arecord. I am testing MIC IN with following command arecord -fdat record.wav when i probe signals I am getting proper MCLK, WCLK(48khz) and BCLK(1.5Mhz) but there is no toggling on data line. would be grateful to any help Thanks in advance Cheers Vivek

2 4

[alsa-devel] hw_ptr accuracy with intel8x0 -> PulseAudio issue
by pl bossart 03 Dec '09

03 Dec '09

Hi, I have been having issues for a while now with PulseAudio running on an older Thinkpad X41. PulseAudio relies on snd_pcm_avail() to detect underflows and increases the timer watermark when they happen. There are actually no audible underflows though... So I instrumented both PulseAudio (enabling DEBUG_TIMING in the alsa-sink.c code) and the kernel (latest alsa-kernel git) by adding the following printk in pcm.h: static inline snd_pcm_uframes_t snd_pcm_playback_avail(struct snd_pcm_runtime *runtime) { snd_pcm_sframes_t avail = runtime->status->hw_ptr + runtime->buffer_size - runtime->control->appl_ptr; if (avail < 0) avail += runtime->boundary; else if ((snd_pcm_uframes_t) avail >= runtime->boundary) avail -= runtime->boundary; //plb if(avail > runtime->buffer_size) { snd_printk(KERN_ERR "avail %d hw_ptr %d buffer %d appl_ptr %d \n", avail, runtime->status->hw_ptr, runtime->buffer_size, runtime->control->appl_ptr); } return avail; } In my tests this error condition where avail is larger than the ring buffer happens fairly often. The dmesg log show messages such as [ 834.524479] ALSA include/sound/pcm.h:600: avail 16403 hw_ptr 24351 buffer 16384 appl_ptr 24332 [ 835.266937] ALSA include/sound/pcm.h:600: avail 16407 hw_ptr 55841 buffer 16384 appl_ptr 55818 and these messages are correlated with the times when PulseAudio detects underflows. The full log can be found at http://pastebin.com/f210aa108. I believe the issue is the accuracy of the hw_ptr reported by snd_intel8x0_pcm_pointer. If indeed such a condition occurred, it should be flagged as underflow, shouldn't it? There's been some changes in this part of the code to avoid extremely large values, here the issue seems more subtle, the pointer is only slightly off in most cases. This does screw-up PulseAudio though, the timer watermark keeps increasing and low-latency isn't possible. I'd be more than happy to run more experiments, however I have no idea what the hardware does to report the position of the hw pointer. Thanks for your feedback - Pierre

2 1

[alsa-devel] [PATCH] opti9xx: remove snd_opti9xx fields
by Krzysztof Helt 03 Dec '09

03 Dec '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Remove snd_opti9xx fields which are indirect arguments to the snd_opti9xx_configure(). Pass these values as function arguments. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This patch prepares a driver for a better fix to the irq problem. sound/isa/opti9xx/opti92x-ad1848.c | 110 ++++++++++++++---------------------- 1 files changed, 43 insertions(+), 67 deletions(-) diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 5cd5553..d08c389 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -141,15 +141,7 @@ struct snd_opti9xx { spinlock_t lock; - long wss_base; int irq; - int dma1; - int dma2; - - long fm_port; - - long mpu_port; - int mpu_irq; #ifdef CONFIG_PNP struct pnp_dev *dev; @@ -216,13 +208,7 @@ static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip, spin_lock_init(&chip->lock); - chip->wss_base = -1; chip->irq = -1; - chip->dma1 = -1; - chip->dma2 = -1; - chip->fm_port = -1; - chip->mpu_port = -1; - chip->mpu_irq = -1; switch (hardware) { #ifndef OPTi93X @@ -348,7 +334,10 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, (snd_opti9xx_read(chip, reg) & ~(mask)) | ((value) & (mask))) -static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip) +static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip, + long wss_base, + int irq, int dma1, int dma2, + long mpu_port, int mpu_irq) { unsigned char wss_base_bits; unsigned char irq_bits; @@ -416,7 +405,7 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip) return -EINVAL; } - switch (chip->wss_base) { + switch (wss_base) { case 0x530: wss_base_bits = 0x00; break; @@ -430,14 +419,13 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip) wss_base_bits = 0x02; break; default: - snd_printk(KERN_WARNING "WSS port 0x%lx not valid\n", - chip->wss_base); + snd_printk(KERN_WARNING "WSS port 0x%lx not valid\n", wss_base); goto __skip_base; } snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(1), wss_base_bits << 4, 0x30); __skip_base: - switch (chip->irq) { + switch (irq) { //#ifdef OPTi93X case 5: irq_bits = 0x05; @@ -456,11 +444,11 @@ __skip_base: irq_bits = 0x04; break; default: - snd_printk(KERN_WARNING "WSS irq # %d not valid\n", chip->irq); + snd_printk(KERN_WARNING "WSS irq # %d not valid\n", irq); goto __skip_resources; } - switch (chip->dma1) { + switch (dma1) { case 0: dma_bits = 0x01; break; @@ -471,38 +459,36 @@ __skip_base: dma_bits = 0x03; break; default: - snd_printk(KERN_WARNING "WSS dma1 # %d not valid\n", - chip->dma1); + snd_printk(KERN_WARNING "WSS dma1 # %d not valid\n", dma1); goto __skip_resources; } #if defined(CS4231) || defined(OPTi93X) - if (chip->dma1 == chip->dma2) { + if (dma1 == dma2) { snd_printk(KERN_ERR "don't want to share dmas\n"); return -EBUSY; } - switch (chip->dma2) { + switch (dma2) { case 0: case 1: break; default: - snd_printk(KERN_WARNING "WSS dma2 # %d not valid\n", - chip->dma2); + snd_printk(KERN_WARNING "WSS dma2 # %d not valid\n", dma2); goto __skip_resources; } dma_bits |= 0x04; #endif /* CS4231 || OPTi93X */ #ifndef OPTi93X - outb(irq_bits << 3 | dma_bits, chip->wss_base); + outb(irq_bits << 3 | dma_bits, wss_base); #else /* OPTi93X */ snd_opti9xx_write(chip, OPTi9XX_MC_REG(3), (irq_bits << 3 | dma_bits)); #endif /* OPTi93X */ __skip_resources: if (chip->hardware > OPTi9XX_HW_82C928) { - switch (chip->mpu_port) { + switch (mpu_port) { case 0: case -1: break; @@ -520,12 +506,11 @@ __skip_resources: break; default: snd_printk(KERN_WARNING - "MPU-401 port 0x%lx not valid\n", - chip->mpu_port); + "MPU-401 port 0x%lx not valid\n", mpu_port); goto __skip_mpu; } - switch (chip->mpu_irq) { + switch (mpu_irq) { case 5: mpu_irq_bits = 0x02; break; @@ -540,12 +525,12 @@ __skip_resources: break; default: snd_printk(KERN_WARNING "MPU-401 irq # %d not valid\n", - chip->mpu_irq); + mpu_irq); goto __skip_mpu; } snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), - (chip->mpu_port <= 0) ? 0x00 : + (mpu_port <= 0) ? 0x00 : 0x80 | mpu_port_bits << 5 | mpu_irq_bits << 3, 0xf8); } @@ -701,6 +686,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) { static long possible_ports[] = {0x530, 0xe80, 0xf40, 0x604, -1}; int error; + int xdma2; struct snd_opti9xx *chip = card->private_data; struct snd_wss *codec; #ifdef CS4231 @@ -715,31 +701,25 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) "OPTi9xx MC")) == NULL) return -ENOMEM; - chip->wss_base = port; - chip->fm_port = fm_port; - chip->mpu_port = mpu_port; - chip->irq = irq; - chip->mpu_irq = mpu_irq; - chip->dma1 = dma1; #if defined(CS4231) || defined(OPTi93X) - chip->dma2 = dma2; + xdma2 = dma2; #else - chip->dma2 = -1; + xdma2 = -1; #endif - if (chip->wss_base == SNDRV_AUTO_PORT) { - chip->wss_base = snd_legacy_find_free_ioport(possible_ports, 4); - if (chip->wss_base < 0) { + if (port == SNDRV_AUTO_PORT) { + port = snd_legacy_find_free_ioport(possible_ports, 4); + if (port < 0) { snd_printk(KERN_ERR "unable to find a free WSS port\n"); return -EBUSY; } } - error = snd_opti9xx_configure(chip); + error = snd_opti9xx_configure(chip, port, irq, dma1, xdma2, + mpu_port, mpu_irq); if (error) return error; - error = snd_wss_create(card, chip->wss_base + 4, -1, - chip->irq, chip->dma1, chip->dma2, + error = snd_wss_create(card, port + 4, -1, irq, dma1, xdma2, #ifdef OPTi93X WSS_HW_OPTI93X, WSS_HWSHARE_IRQ, #else @@ -763,35 +743,35 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) return error; #endif #ifdef OPTi93X - error = request_irq(chip->irq, snd_opti93x_interrupt, + error = request_irq(irq, snd_opti93x_interrupt, IRQF_DISABLED, DEV_NAME" - WSS", codec); if (error < 0) { snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", chip->irq); return error; } #endif + chip->irq = irq; strcpy(card->driver, chip->name); sprintf(card->shortname, "OPTi %s", card->driver); #if defined(CS4231) || defined(OPTi93X) sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d&%d", - card->shortname, pcm->name, chip->wss_base + 4, - chip->irq, chip->dma1, chip->dma2); + card->shortname, pcm->name, port + 4, irq, dma1, xdma2); #else sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d", - card->shortname, pcm->name, chip->wss_base + 4, - chip->irq, chip->dma1); + card->shortname, pcm->name, port + 4, irq, dma1); #endif /* CS4231 || OPTi93X */ - if (chip->mpu_port <= 0 || chip->mpu_port == SNDRV_AUTO_PORT) + if (mpu_port <= 0 || mpu_port == SNDRV_AUTO_PORT) rmidi = NULL; - else - if ((error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, - chip->mpu_port, 0, chip->mpu_irq, IRQF_DISABLED, - &rmidi))) + else { + error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, + mpu_port, 0, mpu_irq, IRQF_DISABLED, &rmidi); + if (error) snd_printk(KERN_WARNING "no MPU-401 device at 0x%lx?\n", - chip->mpu_port); + mpu_port); + } - if (chip->fm_port > 0 && chip->fm_port != SNDRV_AUTO_PORT) { + if (fm_port > 0 && fm_port != SNDRV_AUTO_PORT) { struct snd_opl3 *opl3 = NULL; #ifndef OPTi93X if (chip->hardware == OPTi9XX_HW_82C928 || @@ -801,9 +781,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) /* assume we have an OPL4 */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(2), 0x20, 0x20); - if (snd_opl4_create(card, - chip->fm_port, - chip->fm_port - 8, + if (snd_opl4_create(card, fm_port, fm_port - 8, 2, &opl3, &opl4) < 0) { /* no luck, use OPL3 instead */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(2), @@ -811,12 +789,10 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) } } #endif /* !OPTi93X */ - if (!opl3 && snd_opl3_create(card, - chip->fm_port, - chip->fm_port + 2, + if (!opl3 && snd_opl3_create(card, fm_port, fm_port + 2, OPL3_HW_AUTO, 0, &opl3) < 0) { snd_printk(KERN_WARNING "no OPL device at 0x%lx-0x%lx\n", - chip->fm_port, chip->fm_port + 4 - 1); + fm_port, fm_port + 4 - 1); } if (opl3) { error = snd_opl3_hwdep_new(opl3, 0, 1, &synth); -- 1.6.4 ---------------------------------------------------------------------- Pobierz i graj w swoim telefonie. Sprawdz >>> http://link.interia.pl/f24c1

2 1

[alsa-devel] [PATCH]PM_QOS-to-use-handle-based-requests-pcm-update 5/5
by mark gross 02 Dec '09

02 Dec '09

This is the patch to update sound pcm's dependencies on the pmqos api that was changed in the first patch in this series. Updated per input from RFC. --mgross Signed-off-by: mark gross <mgross(a)linux.intel.com>

2 2

[alsa-devel] [PATCH 1/2] opti93x: fix irq releasing if the irq cannot be allocated
by Krzysztof Helt 02 Dec '09

02 Dec '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Move irq allocating earlier so the codec pointer and codec->irq are not set if the request_irq() fails. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The third version. The irq is correctly freed if the snd_wss_create() fails (this is the only change since the previous version). sound/isa/opti9xx/opti92x-ad1848.c | 31 +++++++++++++++++++------------ 1 files changed, 19 insertions(+), 12 deletions(-) diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 5cd5553..7c97ca8 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -558,10 +558,13 @@ __skip_mpu: static irqreturn_t snd_opti93x_interrupt(int irq, void *dev_id) { - struct snd_wss *codec = dev_id; - struct snd_opti9xx *chip = codec->card->private_data; + struct snd_opti9xx *chip = dev_id; + struct snd_wss *codec = chip->codec; unsigned char status; + if (!codec) + return IRQ_HANDLED; + status = snd_opti9xx_read(chip, OPTi9XX_MC_REG(11)); if ((status & OPTi93X_IRQ_PLAYBACK) && codec->playback_substream) snd_pcm_period_elapsed(codec->playback_substream); @@ -690,7 +693,7 @@ static void snd_card_opti9xx_free(struct snd_card *card) struct snd_wss *codec = chip->codec; if (codec && codec->irq > 0) { disable_irq(codec->irq); - free_irq(codec->irq, codec); + free_irq(codec->irq, chip); } #endif release_and_free_resource(chip->res_mc_base); @@ -738,6 +741,14 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) if (error) return error; +#ifdef OPTi93X + error = request_irq(chip->irq, snd_opti93x_interrupt, + IRQF_DISABLED, DEV_NAME" - WSS", chip); + if (error < 0) { + snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", chip->irq); + return error; + } +#endif error = snd_wss_create(card, chip->wss_base + 4, -1, chip->irq, chip->dma1, chip->dma2, #ifdef OPTi93X @@ -746,8 +757,12 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) WSS_HW_DETECT, 0, #endif &codec); - if (error < 0) + if (error < 0) { +#ifdef OPTi93X + free_irq(chip->irq, chip); +#endif return error; + } #ifdef OPTi93X chip->codec = codec; #endif @@ -762,14 +777,6 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) if (error < 0) return error; #endif -#ifdef OPTi93X - error = request_irq(chip->irq, snd_opti93x_interrupt, - IRQF_DISABLED, DEV_NAME" - WSS", codec); - if (error < 0) { - snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", chip->irq); - return error; - } -#endif strcpy(card->driver, chip->name); sprintf(card->shortname, "OPTi %s", card->driver); #if defined(CS4231) || defined(OPTi93X) -- 1.6.4 ---------------------------------------------------------------------- Jak zrobi� swoj� w�asn� tapet� na telefon? Sprawdz >>> http://link.interia.pl/f24c3

2 1

[alsa-devel] [PATCH 2/2] opti9xx: remove snd_opti9xx fields
by Krzysztof Helt 02 Dec '09

02 Dec '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Remove snd_opti9xx fields which are indirect arguments to the snd_opti9xx_configure(). Pass these values as function arguments. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/opti9xx/opti92x-ad1848.c | 118 ++++++++++++++---------------------- 1 files changed, 45 insertions(+), 73 deletions(-) diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 7c97ca8..a4247f7 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -141,16 +141,6 @@ struct snd_opti9xx { spinlock_t lock; - long wss_base; - int irq; - int dma1; - int dma2; - - long fm_port; - - long mpu_port; - int mpu_irq; - #ifdef CONFIG_PNP struct pnp_dev *dev; struct pnp_dev *devmpu; @@ -216,14 +206,6 @@ static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip, spin_lock_init(&chip->lock); - chip->wss_base = -1; - chip->irq = -1; - chip->dma1 = -1; - chip->dma2 = -1; - chip->fm_port = -1; - chip->mpu_port = -1; - chip->mpu_irq = -1; - switch (hardware) { #ifndef OPTi93X case OPTi9XX_HW_82C928: @@ -348,7 +330,10 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, (snd_opti9xx_read(chip, reg) & ~(mask)) | ((value) & (mask))) -static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip) +static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip, + long wss_base, + int irq, int dma1, int dma2, + long mpu_port, int mpu_irq) { unsigned char wss_base_bits; unsigned char irq_bits; @@ -416,7 +401,7 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip) return -EINVAL; } - switch (chip->wss_base) { + switch (wss_base) { case 0x530: wss_base_bits = 0x00; break; @@ -430,14 +415,13 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip) wss_base_bits = 0x02; break; default: - snd_printk(KERN_WARNING "WSS port 0x%lx not valid\n", - chip->wss_base); + snd_printk(KERN_WARNING "WSS port 0x%lx not valid\n", wss_base); goto __skip_base; } snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(1), wss_base_bits << 4, 0x30); __skip_base: - switch (chip->irq) { + switch (irq) { //#ifdef OPTi93X case 5: irq_bits = 0x05; @@ -456,11 +440,11 @@ __skip_base: irq_bits = 0x04; break; default: - snd_printk(KERN_WARNING "WSS irq # %d not valid\n", chip->irq); + snd_printk(KERN_WARNING "WSS irq # %d not valid\n", irq); goto __skip_resources; } - switch (chip->dma1) { + switch (dma1) { case 0: dma_bits = 0x01; break; @@ -471,38 +455,36 @@ __skip_base: dma_bits = 0x03; break; default: - snd_printk(KERN_WARNING "WSS dma1 # %d not valid\n", - chip->dma1); + snd_printk(KERN_WARNING "WSS dma1 # %d not valid\n", dma1); goto __skip_resources; } #if defined(CS4231) || defined(OPTi93X) - if (chip->dma1 == chip->dma2) { + if (dma1 == dma2) { snd_printk(KERN_ERR "don't want to share dmas\n"); return -EBUSY; } - switch (chip->dma2) { + switch (dma2) { case 0: case 1: break; default: - snd_printk(KERN_WARNING "WSS dma2 # %d not valid\n", - chip->dma2); + snd_printk(KERN_WARNING "WSS dma2 # %d not valid\n", dma2); goto __skip_resources; } dma_bits |= 0x04; #endif /* CS4231 || OPTi93X */ #ifndef OPTi93X - outb(irq_bits << 3 | dma_bits, chip->wss_base); + outb(irq_bits << 3 | dma_bits, wss_base); #else /* OPTi93X */ snd_opti9xx_write(chip, OPTi9XX_MC_REG(3), (irq_bits << 3 | dma_bits)); #endif /* OPTi93X */ __skip_resources: if (chip->hardware > OPTi9XX_HW_82C928) { - switch (chip->mpu_port) { + switch (mpu_port) { case 0: case -1: break; @@ -520,12 +502,11 @@ __skip_resources: break; default: snd_printk(KERN_WARNING - "MPU-401 port 0x%lx not valid\n", - chip->mpu_port); + "MPU-401 port 0x%lx not valid\n", mpu_port); goto __skip_mpu; } - switch (chip->mpu_irq) { + switch (mpu_irq) { case 5: mpu_irq_bits = 0x02; break; @@ -540,12 +521,12 @@ __skip_resources: break; default: snd_printk(KERN_WARNING "MPU-401 irq # %d not valid\n", - chip->mpu_irq); + mpu_irq); goto __skip_mpu; } snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), - (chip->mpu_port <= 0) ? 0x00 : + (mpu_port <= 0) ? 0x00 : 0x80 | mpu_port_bits << 5 | mpu_irq_bits << 3, 0xf8); } @@ -704,6 +685,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) { static long possible_ports[] = {0x530, 0xe80, 0xf40, 0x604, -1}; int error; + int xdma2; struct snd_opti9xx *chip = card->private_data; struct snd_wss *codec; #ifdef CS4231 @@ -718,39 +700,33 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) "OPTi9xx MC")) == NULL) return -ENOMEM; - chip->wss_base = port; - chip->fm_port = fm_port; - chip->mpu_port = mpu_port; - chip->irq = irq; - chip->mpu_irq = mpu_irq; - chip->dma1 = dma1; #if defined(CS4231) || defined(OPTi93X) - chip->dma2 = dma2; + xdma2 = dma2; #else - chip->dma2 = -1; + xdma2 = -1; #endif - if (chip->wss_base == SNDRV_AUTO_PORT) { - chip->wss_base = snd_legacy_find_free_ioport(possible_ports, 4); - if (chip->wss_base < 0) { + if (port == SNDRV_AUTO_PORT) { + port = snd_legacy_find_free_ioport(possible_ports, 4); + if (port < 0) { snd_printk(KERN_ERR "unable to find a free WSS port\n"); return -EBUSY; } } - error = snd_opti9xx_configure(chip); + error = snd_opti9xx_configure(chip, port, irq, dma1, xdma2, + mpu_port, mpu_irq); if (error) return error; #ifdef OPTi93X - error = request_irq(chip->irq, snd_opti93x_interrupt, + error = request_irq(irq, snd_opti93x_interrupt, IRQF_DISABLED, DEV_NAME" - WSS", chip); if (error < 0) { - snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", chip->irq); + snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", irq); return error; } #endif - error = snd_wss_create(card, chip->wss_base + 4, -1, - chip->irq, chip->dma1, chip->dma2, + error = snd_wss_create(card, port + 4, -1, irq, dma1, xdma2, #ifdef OPTi93X WSS_HW_OPTI93X, WSS_HWSHARE_IRQ, #else @@ -759,7 +735,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) &codec); if (error < 0) { #ifdef OPTi93X - free_irq(chip->irq, chip); + free_irq(irq, chip); #endif return error; } @@ -781,24 +757,24 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) sprintf(card->shortname, "OPTi %s", card->driver); #if defined(CS4231) || defined(OPTi93X) sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d&%d", - card->shortname, pcm->name, chip->wss_base + 4, - chip->irq, chip->dma1, chip->dma2); + card->shortname, pcm->name, port + 4, irq, dma1, xdma2); #else sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d", - card->shortname, pcm->name, chip->wss_base + 4, - chip->irq, chip->dma1); + card->shortname, pcm->name, port + 4, irq, dma1); #endif /* CS4231 || OPTi93X */ - if (chip->mpu_port <= 0 || chip->mpu_port == SNDRV_AUTO_PORT) + if (mpu_port <= 0 || mpu_port == SNDRV_AUTO_PORT) rmidi = NULL; - else - if ((error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, - chip->mpu_port, 0, chip->mpu_irq, IRQF_DISABLED, - &rmidi))) + else { + error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, + mpu_port, 0, mpu_irq, IRQF_DISABLED, + &rmidi); + if (error) snd_printk(KERN_WARNING "no MPU-401 device at 0x%lx?\n", - chip->mpu_port); + mpu_port); + } - if (chip->fm_port > 0 && chip->fm_port != SNDRV_AUTO_PORT) { + if (fm_port > 0 && fm_port != SNDRV_AUTO_PORT) { struct snd_opl3 *opl3 = NULL; #ifndef OPTi93X if (chip->hardware == OPTi9XX_HW_82C928 || @@ -808,9 +784,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) /* assume we have an OPL4 */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(2), 0x20, 0x20); - if (snd_opl4_create(card, - chip->fm_port, - chip->fm_port - 8, + if (snd_opl4_create(card, fm_port, fm_port - 8, 2, &opl3, &opl4) < 0) { /* no luck, use OPL3 instead */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(2), @@ -818,12 +792,10 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) } } #endif /* !OPTi93X */ - if (!opl3 && snd_opl3_create(card, - chip->fm_port, - chip->fm_port + 2, + if (!opl3 && snd_opl3_create(card, fm_port, fm_port + 2, OPL3_HW_AUTO, 0, &opl3) < 0) { snd_printk(KERN_WARNING "no OPL device at 0x%lx-0x%lx\n", - chip->fm_port, chip->fm_port + 4 - 1); + fm_port, fm_port + 4 - 1); } if (opl3) { error = snd_opl3_hwdep_new(opl3, 0, 1, &synth); -- 1.6.4 ---------------------------------------------------------------------- Do�aduj sw�j telefon! 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1 0

[alsa-devel] rewind operations in pcm_ops
by Singaravelan Nallasellan 02 Dec '09

02 Dec '09

Hi, The audio subsystem in platform has its own buffering. It could buffer upto 256KB. Our driver uses indirect callback to send the buffers and sizes to audio subsystem whenever the "ack" is called. If rewind is required during the playback, ALSA pcm native interface is not calling the driver operations for rewind. I am planning to add rewind operations to pcm_ops and make changes to pcm files. Please suggest me whether this is acceptable. Thanks -Sing

1 0

[PATCH 5/5] PM_QOS to use handle based requests pcm update
by mark gross 01 Dec '09

01 Dec '09

--- include/sound/pcm.h | 3 ++- sound/core/pcm.c | 3 --- sound/core/pcm_native.c | 14 ++++++++------ 3 files changed, 10 insertions(+), 10 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index de6d981..91daac3 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -29,6 +29,7 @@ #include <linux/poll.h> #include <linux/mm.h> #include <linux/bitops.h> +#include <linux/pm_qos_params.h> #define snd_pcm_substream_chip(substream) ((substream)->private_data) #define snd_pcm_chip(pcm) ((pcm)->private_data) @@ -355,7 +356,7 @@ struct snd_pcm_substream { int number; char name[32]; /* substream name */ int stream; /* stream (direction) */ - char latency_id[20]; /* latency identifier */ + struct pm_qos_request_list *latency_pm_qos_req; /* pm_qos latency request */ size_t buffer_bytes_max; /* limit ring buffer size */ struct snd_dma_buffer dma_buffer; unsigned int dma_buf_id; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index c69c60b..6faf4b5 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -647,9 +647,6 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) substream->number = idx; substream->stream = stream; sprintf(substream->name, "subdevice #%i", idx); - snprintf(substream->latency_id, sizeof(substream->latency_id), - "ALSA-PCM%d-%d%c%d", pcm->card->number, pcm->device, - (stream ? 'c' : 'p'), idx); substream->buffer_bytes_max = UINT_MAX; if (prev == NULL) pstr->substream = substream; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index ab73edf..2aab6cf 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -447,11 +447,13 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, snd_pcm_timer_resolution_change(substream); runtime->status->state = SNDRV_PCM_STATE_SETUP; - pm_qos_remove_requirement(PM_QOS_CPU_DMA_LATENCY, - substream->latency_id); + if (substream->latency_pm_qos_req) { + pm_qos_remove_request(substream->latency_pm_qos_req); + substream->latency_pm_qos_req = NULL; + } if ((usecs = period_to_usecs(runtime)) >= 0) - pm_qos_add_requirement(PM_QOS_CPU_DMA_LATENCY, - substream->latency_id, usecs); + substream->latency_pm_qos_req = pm_qos_add_request( + PM_QOS_CPU_DMA_LATENCY, usecs); return 0; _error: /* hardware might be unuseable from this time, @@ -506,8 +508,8 @@ static int snd_pcm_hw_free(struct snd_pcm_substream *substream) if (substream->ops->hw_free) result = substream->ops->hw_free(substream); runtime->status->state = SNDRV_PCM_STATE_OPEN; - pm_qos_remove_requirement(PM_QOS_CPU_DMA_LATENCY, - substream->latency_id); + pm_qos_remove_request(substream->latency_pm_qos_req); + substream->latency_pm_qos_req = NULL; return result; } -- 1.6.3.3

1 0

[alsa-devel] [PATCH] Fix mixer map for Hercules Gamesurround Muse Pocket LT
by Alexey Fisher 01 Dec '09

01 Dec '09

Muse Pocket use brocken mixer names, so alsamixer and PA can't use it correctly This patch add quirk to overwirte default mixers. Signed-off-by: Alexey Fisher <bug-track(a)fisher-privat.net> modified: sound/usb/usbmixer_maps.c --- sound/usb/usbmixer_maps.c | 23 +++++++++++++++++++++++ 1 files changed, 23 insertions(+), 0 deletions(-) diff --git a/sound/usb/usbmixer_maps.c b/sound/usb/usbmixer_maps.c index 3e5d66c..77c3588 100644 --- a/sound/usb/usbmixer_maps.c +++ b/sound/usb/usbmixer_maps.c @@ -277,6 +277,22 @@ static struct usbmix_name_map scratch_live_map[] = { { 0 } /* terminator */ }; +/* "Gamesurround Muse Pocket LT" looks same like "Sound Blaster MP3+" + * most importand difference is SU[8], it should be set to "Capture Source" + * to make alsamixer and PA working properly. + * FIXME: or mp3plus_map should use "Capture Source" too, + * so this maps can be merget + */ +static struct usbmix_name_map hercules_usb51_map[] = { + { 8, "Capture Source" }, /* SU, default "PCM Capture Source" */ + { 9, "Master Playback" }, /* FU, default "Speaker Playback" */ + { 10, "Mic Boost", 7 }, /* FU, default "Auto Gain Input" */ + { 11, "Line Capture" }, /* FU, default "PCM Capture" */ + { 13, "Mic Bypass Playback" }, /* FU, default "Mic Playback" */ + { 14, "Line Bypass Playback" }, /* FU, default "Line Playback" */ + { 0 } /* terminator */ +}; + /* * Control map entries */ @@ -316,6 +332,13 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .ignore_ctl_error = 1, }, { + /* Hercules Gamesurround Muse Pocket LT + * (USB 5.1 Channel Audio Adapter) + */ + .id = USB_ID(0x06f8, 0xc000), + .map = hercules_usb51_map, + }, + { .id = USB_ID(0x08bb, 0x2702), .map = linex_map, .ignore_ctl_error = 1, -- 1.6.3.3

2 1

[alsa-devel] [PATCH 1/2] opti93x: fix irq releasing if the irq cannot be allocated
by Krzysztof Helt 01 Dec '09

01 Dec '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Fix the bug that an irq is released if the irq's allocation fails. Move irq allocation earlier so the codec->irq is not set if the request_irq() fails. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- A different approach than the previous one. sound/isa/opti9xx/opti92x-ad1848.c | 25 ++++++++++++++----------- 1 files changed, 14 insertions(+), 11 deletions(-) diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 5cd5553..fd32001 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -558,10 +558,13 @@ __skip_mpu: static irqreturn_t snd_opti93x_interrupt(int irq, void *dev_id) { - struct snd_wss *codec = dev_id; - struct snd_opti9xx *chip = codec->card->private_data; + struct snd_opti9xx *chip = dev_id; + struct snd_wss *codec = chip->codec; unsigned char status; + if (!codec) + return IRQ_HANDLED; + status = snd_opti9xx_read(chip, OPTi9XX_MC_REG(11)); if ((status & OPTi93X_IRQ_PLAYBACK) && codec->playback_substream) snd_pcm_period_elapsed(codec->playback_substream); @@ -690,7 +693,7 @@ static void snd_card_opti9xx_free(struct snd_card *card) struct snd_wss *codec = chip->codec; if (codec && codec->irq > 0) { disable_irq(codec->irq); - free_irq(codec->irq, codec); + free_irq(codec->irq, chip); } #endif release_and_free_resource(chip->res_mc_base); @@ -738,6 +741,14 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) if (error) return error; +#ifdef OPTi93X + error = request_irq(chip->irq, snd_opti93x_interrupt, + IRQF_DISABLED, DEV_NAME" - WSS", chip); + if (error < 0) { + snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", chip->irq); + return error; + } +#endif error = snd_wss_create(card, chip->wss_base + 4, -1, chip->irq, chip->dma1, chip->dma2, #ifdef OPTi93X @@ -762,14 +773,6 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) if (error < 0) return error; #endif -#ifdef OPTi93X - error = request_irq(chip->irq, snd_opti93x_interrupt, - IRQF_DISABLED, DEV_NAME" - WSS", codec); - if (error < 0) { - snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", chip->irq); - return error; - } -#endif strcpy(card->driver, chip->name); sprintf(card->shortname, "OPTi %s", card->driver); #if defined(CS4231) || defined(OPTi93X) -- 1.6.4 ---------------------------------------------------------------------- KONKURS! Kliknij i wygraj luksusowe auto na weekend lub jedn� z 650 innych nagr�d od L’Oreal Men Expert!! >> http://link.interia.pl/f2428

2 2

[alsa-devel] [PATCH 2/2] opti9xx: remove snd_opti9xx fields
by Krzysztof Helt 30 Nov '09

30 Nov '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Remove snd_opti9xx fields which are indirect arguments to the snd_opti9xx_configure(). Pass these values as function arguments. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/opti9xx/opti92x-ad1848.c | 116 ++++++++++++++---------------------- 1 files changed, 44 insertions(+), 72 deletions(-) diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index fd32001..1392d3c 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -141,16 +141,6 @@ struct snd_opti9xx { spinlock_t lock; - long wss_base; - int irq; - int dma1; - int dma2; - - long fm_port; - - long mpu_port; - int mpu_irq; - #ifdef CONFIG_PNP struct pnp_dev *dev; struct pnp_dev *devmpu; @@ -216,14 +206,6 @@ static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip, spin_lock_init(&chip->lock); - chip->wss_base = -1; - chip->irq = -1; - chip->dma1 = -1; - chip->dma2 = -1; - chip->fm_port = -1; - chip->mpu_port = -1; - chip->mpu_irq = -1; - switch (hardware) { #ifndef OPTi93X case OPTi9XX_HW_82C928: @@ -348,7 +330,10 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, (snd_opti9xx_read(chip, reg) & ~(mask)) | ((value) & (mask))) -static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip) +static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip, + long wss_base, + int irq, int dma1, int dma2, + long mpu_port, int mpu_irq) { unsigned char wss_base_bits; unsigned char irq_bits; @@ -416,7 +401,7 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip) return -EINVAL; } - switch (chip->wss_base) { + switch (wss_base) { case 0x530: wss_base_bits = 0x00; break; @@ -430,14 +415,13 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip) wss_base_bits = 0x02; break; default: - snd_printk(KERN_WARNING "WSS port 0x%lx not valid\n", - chip->wss_base); + snd_printk(KERN_WARNING "WSS port 0x%lx not valid\n", wss_base); goto __skip_base; } snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(1), wss_base_bits << 4, 0x30); __skip_base: - switch (chip->irq) { + switch (irq) { //#ifdef OPTi93X case 5: irq_bits = 0x05; @@ -456,11 +440,11 @@ __skip_base: irq_bits = 0x04; break; default: - snd_printk(KERN_WARNING "WSS irq # %d not valid\n", chip->irq); + snd_printk(KERN_WARNING "WSS irq # %d not valid\n", irq); goto __skip_resources; } - switch (chip->dma1) { + switch (dma1) { case 0: dma_bits = 0x01; break; @@ -471,38 +455,36 @@ __skip_base: dma_bits = 0x03; break; default: - snd_printk(KERN_WARNING "WSS dma1 # %d not valid\n", - chip->dma1); + snd_printk(KERN_WARNING "WSS dma1 # %d not valid\n", dma1); goto __skip_resources; } #if defined(CS4231) || defined(OPTi93X) - if (chip->dma1 == chip->dma2) { + if (dma1 == dma2) { snd_printk(KERN_ERR "don't want to share dmas\n"); return -EBUSY; } - switch (chip->dma2) { + switch (dma2) { case 0: case 1: break; default: - snd_printk(KERN_WARNING "WSS dma2 # %d not valid\n", - chip->dma2); + snd_printk(KERN_WARNING "WSS dma2 # %d not valid\n", dma2); goto __skip_resources; } dma_bits |= 0x04; #endif /* CS4231 || OPTi93X */ #ifndef OPTi93X - outb(irq_bits << 3 | dma_bits, chip->wss_base); + outb(irq_bits << 3 | dma_bits, wss_base); #else /* OPTi93X */ snd_opti9xx_write(chip, OPTi9XX_MC_REG(3), (irq_bits << 3 | dma_bits)); #endif /* OPTi93X */ __skip_resources: if (chip->hardware > OPTi9XX_HW_82C928) { - switch (chip->mpu_port) { + switch (mpu_port) { case 0: case -1: break; @@ -520,12 +502,11 @@ __skip_resources: break; default: snd_printk(KERN_WARNING - "MPU-401 port 0x%lx not valid\n", - chip->mpu_port); + "MPU-401 port 0x%lx not valid\n", mpu_port); goto __skip_mpu; } - switch (chip->mpu_irq) { + switch (mpu_irq) { case 5: mpu_irq_bits = 0x02; break; @@ -540,12 +521,12 @@ __skip_resources: break; default: snd_printk(KERN_WARNING "MPU-401 irq # %d not valid\n", - chip->mpu_irq); + mpu_irq); goto __skip_mpu; } snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), - (chip->mpu_port <= 0) ? 0x00 : + (mpu_port <= 0) ? 0x00 : 0x80 | mpu_port_bits << 5 | mpu_irq_bits << 3, 0xf8); } @@ -704,6 +685,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) { static long possible_ports[] = {0x530, 0xe80, 0xf40, 0x604, -1}; int error; + int xdma2; struct snd_opti9xx *chip = card->private_data; struct snd_wss *codec; #ifdef CS4231 @@ -718,39 +700,33 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) "OPTi9xx MC")) == NULL) return -ENOMEM; - chip->wss_base = port; - chip->fm_port = fm_port; - chip->mpu_port = mpu_port; - chip->irq = irq; - chip->mpu_irq = mpu_irq; - chip->dma1 = dma1; #if defined(CS4231) || defined(OPTi93X) - chip->dma2 = dma2; + xdma2 = dma2; #else - chip->dma2 = -1; + xdma2 = -1; #endif - if (chip->wss_base == SNDRV_AUTO_PORT) { - chip->wss_base = snd_legacy_find_free_ioport(possible_ports, 4); - if (chip->wss_base < 0) { + if (port == SNDRV_AUTO_PORT) { + port = snd_legacy_find_free_ioport(possible_ports, 4); + if (port < 0) { snd_printk(KERN_ERR "unable to find a free WSS port\n"); return -EBUSY; } } - error = snd_opti9xx_configure(chip); + error = snd_opti9xx_configure(chip, port, irq, dma1, xdma2, + mpu_port, mpu_irq); if (error) return error; #ifdef OPTi93X - error = request_irq(chip->irq, snd_opti93x_interrupt, + error = request_irq(irq, snd_opti93x_interrupt, IRQF_DISABLED, DEV_NAME" - WSS", chip); if (error < 0) { - snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", chip->irq); + snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", irq); return error; } #endif - error = snd_wss_create(card, chip->wss_base + 4, -1, - chip->irq, chip->dma1, chip->dma2, + error = snd_wss_create(card, port + 4, -1, irq, dma1, xdma2, #ifdef OPTi93X WSS_HW_OPTI93X, WSS_HWSHARE_IRQ, #else @@ -777,24 +753,24 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) sprintf(card->shortname, "OPTi %s", card->driver); #if defined(CS4231) || defined(OPTi93X) sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d&%d", - card->shortname, pcm->name, chip->wss_base + 4, - chip->irq, chip->dma1, chip->dma2); + card->shortname, pcm->name, port + 4, irq, dma1, xdma2); #else sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d", - card->shortname, pcm->name, chip->wss_base + 4, - chip->irq, chip->dma1); + card->shortname, pcm->name, port + 4, irq, dma1); #endif /* CS4231 || OPTi93X */ - if (chip->mpu_port <= 0 || chip->mpu_port == SNDRV_AUTO_PORT) + if (mpu_port <= 0 || mpu_port == SNDRV_AUTO_PORT) rmidi = NULL; - else - if ((error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, - chip->mpu_port, 0, chip->mpu_irq, IRQF_DISABLED, - &rmidi))) + else { + error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, + mpu_port, 0, mpu_irq, IRQF_DISABLED, + &rmidi); + if (error) snd_printk(KERN_WARNING "no MPU-401 device at 0x%lx?\n", - chip->mpu_port); + mpu_port); + } - if (chip->fm_port > 0 && chip->fm_port != SNDRV_AUTO_PORT) { + if (fm_port > 0 && fm_port != SNDRV_AUTO_PORT) { struct snd_opl3 *opl3 = NULL; #ifndef OPTi93X if (chip->hardware == OPTi9XX_HW_82C928 || @@ -804,9 +780,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) /* assume we have an OPL4 */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(2), 0x20, 0x20); - if (snd_opl4_create(card, - chip->fm_port, - chip->fm_port - 8, + if (snd_opl4_create(card, fm_port, fm_port - 8, 2, &opl3, &opl4) < 0) { /* no luck, use OPL3 instead */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(2), @@ -814,12 +788,10 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) } } #endif /* !OPTi93X */ - if (!opl3 && snd_opl3_create(card, - chip->fm_port, - chip->fm_port + 2, + if (!opl3 && snd_opl3_create(card, fm_port, fm_port + 2, OPL3_HW_AUTO, 0, &opl3) < 0) { snd_printk(KERN_WARNING "no OPL device at 0x%lx-0x%lx\n", - chip->fm_port, chip->fm_port + 4 - 1); + fm_port, fm_port + 4 - 1); } if (opl3) { error = snd_opl3_hwdep_new(opl3, 0, 1, &synth); -- 1.6.4 ---------------------------------------------------------------------- Lawina prezent�w czeka ... Sprawdz >>> http://link.interia.pl/f24a0

1 0

[alsa-devel] ASoC update for 2.6.32
by Mark Brown 30 Nov '09

30 Nov '09

The following changes since commit 50b6bce59d154b5db137907a5c0ed45a4e7a3829: Mark Brown (1): ASoC: Fix suspend with active audio streams are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 This is a bit late and since the breakage in question has been present for rather a long time it's not super urgent to actually get into 2.6.32 but it'd be nice. Anuj Aggarwal (1): ASoC: AIC23: Fixing infinite loop in resume path sound/soc/codecs/tlv320aic23.c | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-)

2 1

[alsa-devel] [PATCH] ALSA: ASoc: pxa/raumfeld: adopt new snd_soc_dai_set_pll() API
by Daniel Mack 30 Nov '09

30 Nov '09

This is needed for recent changes in the for-2.6.33 branch. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/pxa/raumfeld.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 5d7bb8c..a43a16c 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -116,7 +116,7 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, return ret; /* setup the CPU DAI */ - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, clk); + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, clk, clk); if (ret < 0) return ret; @@ -205,7 +205,7 @@ static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream, return ret; /* setup the CPU DAI */ - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, clk); + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, clk, clk); if (ret < 0) return ret; -- 1.6.5.2

2 3

[alsa-devel] [PATCH v3] ASoC: sh: fsi: Add runtime PM support
by Kuninori Morimoto 30 Nov '09

30 Nov '09

This patch add support runtime PM. Driver callbacks for Runtime PM are empty because the device registers are always re-initialized after pm_runtime_get_sync(). The Runtime PM functions replaces the clock framework module stop bit handling in this driver. Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- v2 -> v3 o remove pm_runtime_ignore_children sound/soc/sh/fsi.c | 39 +++++++++++++++++++++++++-------------- 1 files changed, 25 insertions(+), 14 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index e1a3d1a..9c49c11 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -17,7 +17,7 @@ #include <linux/platform_device.h> #include <linux/delay.h> #include <linux/list.h> -#include <linux/clk.h> +#include <linux/pm_runtime.h> #include <linux/io.h> #include <sound/core.h> #include <sound/pcm.h> @@ -105,7 +105,6 @@ struct fsi_priv { struct fsi_master { void __iomem *base; int irq; - struct clk *clk; struct fsi_priv fsia; struct fsi_priv fsib; struct sh_fsi_platform_info *info; @@ -559,7 +558,7 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, int is_master; int ret = 0; - clk_enable(master->clk); + pm_runtime_get_sync(dai->dev); /* CKG1 */ data = is_play ? (1 << 0) : (1 << 4); @@ -674,7 +673,7 @@ static void fsi_dai_shutdown(struct snd_pcm_substream *substream, fsi_irq_disable(fsi, is_play); fsi_clk_ctrl(fsi, 0); - clk_disable(master->clk); + pm_runtime_put_sync(dai->dev); } static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, @@ -872,7 +871,6 @@ EXPORT_SYMBOL_GPL(fsi_soc_platform); static int fsi_probe(struct platform_device *pdev) { struct resource *res; - char clk_name[8]; unsigned int irq; int ret; @@ -903,14 +901,8 @@ static int fsi_probe(struct platform_device *pdev) master->fsia.base = master->base; master->fsib.base = master->base + 0x40; - /* FSI is based on SPU mstp */ - snprintf(clk_name, sizeof(clk_name), "spu%d", pdev->id); - master->clk = clk_get(NULL, clk_name); - if (IS_ERR(master->clk)) { - dev_err(&pdev->dev, "cannot get %s mstp\n", clk_name); - ret = -EIO; - goto exit_iounmap; - } + pm_runtime_enable(&pdev->dev); + pm_runtime_resume(&pdev->dev); fsi_soc_dai[0].dev = &pdev->dev; fsi_soc_dai[1].dev = &pdev->dev; @@ -935,6 +927,7 @@ exit_free_irq: free_irq(irq, master); exit_iounmap: iounmap(master->base); + pm_runtime_disable(&pdev->dev); exit_kfree: kfree(master); master = NULL; @@ -947,7 +940,7 @@ static int fsi_remove(struct platform_device *pdev) snd_soc_unregister_dais(fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai)); snd_soc_unregister_platform(&fsi_soc_platform); - clk_put(master->clk); + pm_runtime_disable(&pdev->dev); free_irq(master->irq, master); @@ -957,9 +950,27 @@ static int fsi_remove(struct platform_device *pdev) return 0; } +static int fsi_runtime_nop(struct device *dev) +{ + /* Runtime PM callback shared between ->runtime_suspend() + * and ->runtime_resume(). Simply returns success. + * + * This driver re-initializes all registers after + * pm_runtime_get_sync() anyway so there is no need + * to save and restore registers here. + */ + return 0; +} + +static struct dev_pm_ops fsi_pm_ops = { + .runtime_suspend = fsi_runtime_nop, + .runtime_resume = fsi_runtime_nop, +}; + static struct platform_driver fsi_driver = { .driver = { .name = "sh_fsi", + .pm = &fsi_pm_ops, }, .probe = fsi_probe, .remove = fsi_remove, -- 1.6.3.3

2 1

[alsa-devel] AC97 slot information
by Russell King - ARM Linux 30 Nov '09

30 Nov '09

When opening an AC'97 codec under ALSA using snd_ac97_pcm_open(), specifying the number of channels desired (2, 4 or 6) is there a way to read back which slots will be in use? Currently, in the AACI driver, I use a lookup table to convert from channels to a slot mask - if this information is already available from ALSA, I'd much rather use it rather than inventing keeping my own solution. Also, should the 'slots' value passed to snd_ac97_pcm_open() be those in pcm->r[].slots or should it be selected from the number of channels?

3 2

[alsa-devel] [PATCH 1/2] opti-miro: separate comon probing code
by Krzysztof Helt 30 Nov '09

30 Nov '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Separate common probing code in order to use it for PnP probing. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/opti9xx/miro.c | 273 +++++++++++++++++++++++++--------------------- 1 files changed, 147 insertions(+), 126 deletions(-) diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index e374869..c67bc3c 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -1142,28 +1142,39 @@ __skip_mpu: return 0; } +static int __devinit snd_miro_opti_check(struct snd_miro *chip) +{ + unsigned char value; + + chip->res_mc_base = request_region(chip->mc_base, chip->mc_base_size, + "OPTi9xx MC"); + if (chip->res_mc_base == NULL) + return -ENOMEM; + + value = snd_miro_read(chip, OPTi9XX_MC_REG(1)); + if (value != 0xff && value != inb(chip->mc_base + OPTi9XX_MC_REG(1))) + if (value == snd_miro_read(chip, OPTi9XX_MC_REG(1))) + return 0; + + release_and_free_resource(chip->res_mc_base); + chip->res_mc_base = NULL; + + return -ENODEV; +} + static int __devinit snd_card_miro_detect(struct snd_card *card, struct snd_miro *chip) { int i, err; - unsigned char value; for (i = OPTi9XX_HW_82C929; i <= OPTi9XX_HW_82C924; i++) { if ((err = snd_miro_init(chip, i)) < 0) return err; - if ((chip->res_mc_base = request_region(chip->mc_base, chip->mc_base_size, "OPTi9xx MC")) == NULL) - continue; - - value = snd_miro_read(chip, OPTi9XX_MC_REG(1)); - if ((value != 0xff) && (value != inb(chip->mc_base + 1))) - if (value == snd_miro_read(chip, OPTi9XX_MC_REG(1))) - return 1; - - release_and_free_resource(chip->res_mc_base); - chip->res_mc_base = NULL; - + err = snd_miro_opti_check(chip); + if (err == 0) + return 1; } return -ENODEV; @@ -1234,151 +1245,69 @@ static void snd_card_miro_free(struct snd_card *card) release_and_free_resource(miro->res_mc_base); } -static int __devinit snd_miro_match(struct device *devptr, unsigned int n) -{ - return 1; -} - -static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) +static int __devinit snd_miro_probe(struct snd_card *card) { - static long possible_ports[] = {0x530, 0xe80, 0xf40, 0x604, -1}; - static long possible_mpu_ports[] = {0x330, 0x300, 0x310, 0x320, -1}; - static int possible_irqs[] = {11, 9, 10, 7, -1}; - static int possible_mpu_irqs[] = {10, 5, 9, 7, -1}; - static int possible_dma1s[] = {3, 1, 0, -1}; - static int possible_dma2s[][2] = {{1,-1}, {0,-1}, {-1,-1}, {0,-1}}; - int error; - struct snd_miro *miro; + struct snd_miro *miro = card->private_data; struct snd_wss *codec; struct snd_timer *timer; - struct snd_card *card; struct snd_pcm *pcm; struct snd_rawmidi *rmidi; - error = snd_card_create(index, id, THIS_MODULE, - sizeof(struct snd_miro), &card); - if (error < 0) - return error; - - card->private_free = snd_card_miro_free; - miro = card->private_data; - - error = snd_card_miro_detect(card, miro); - if (error < 0) { - snd_card_free(card); - snd_printk(KERN_ERR "unable to detect OPTi9xx chip\n"); - return -ENODEV; + if (!miro->res_mc_base) { + miro->res_mc_base = request_region(miro->mc_base, + miro->mc_base_size, + "miro (OPTi9xx MC)"); + if (miro->res_mc_base == NULL) { + snd_printk(KERN_ERR "request for OPTI9xx MC failed\n"); + return -ENOMEM; + } } - if ((error = snd_card_miro_aci_detect(card, miro)) < 0) { + error = snd_card_miro_aci_detect(card, miro); + if (error < 0) { snd_card_free(card); snd_printk(KERN_ERR "unable to detect aci chip\n"); return -ENODEV; } - /* init proc interface */ - snd_miro_proc_init(card, miro); - - - if (! miro->res_mc_base && - (miro->res_mc_base = request_region(miro->mc_base, miro->mc_base_size, - "miro (OPTi9xx MC)")) == NULL) { - snd_card_free(card); - snd_printk(KERN_ERR "request for OPTI9xx MC failed\n"); - return -ENOMEM; - } - miro->wss_base = port; + miro->mpu_port = mpu_port; miro->irq = irq; miro->mpu_irq = mpu_irq; miro->dma1 = dma1; miro->dma2 = dma2; - if (miro->wss_base == SNDRV_AUTO_PORT) { - if ((miro->wss_base = snd_legacy_find_free_ioport(possible_ports, 4)) < 0) { - snd_card_free(card); - snd_printk(KERN_ERR "unable to find a free WSS port\n"); - return -EBUSY; - } - } - - if (mpu_port == SNDRV_AUTO_PORT) { - mpu_port = snd_legacy_find_free_ioport(possible_mpu_ports, 2); - if (mpu_port < 0) { - snd_card_free(card); - snd_printk(KERN_ERR "unable to find a free MPU401 port\n"); - return -EBUSY; - } - } - miro->mpu_port = mpu_port; - - if (miro->irq == SNDRV_AUTO_IRQ) { - if ((miro->irq = snd_legacy_find_free_irq(possible_irqs)) < 0) { - snd_card_free(card); - snd_printk(KERN_ERR "unable to find a free IRQ\n"); - return -EBUSY; - } - } - if (miro->mpu_irq == SNDRV_AUTO_IRQ) { - if ((miro->mpu_irq = snd_legacy_find_free_irq(possible_mpu_irqs)) < 0) { - snd_card_free(card); - snd_printk(KERN_ERR "unable to find a free MPU401 IRQ\n"); - return -EBUSY; - } - } - if (miro->dma1 == SNDRV_AUTO_DMA) { - if ((miro->dma1 = snd_legacy_find_free_dma(possible_dma1s)) < 0) { - snd_card_free(card); - snd_printk(KERN_ERR "unable to find a free DMA1\n"); - return -EBUSY; - } - } - if (miro->dma2 == SNDRV_AUTO_DMA) { - if ((miro->dma2 = snd_legacy_find_free_dma(possible_dma2s[miro->dma1 % 4])) < 0) { - snd_card_free(card); - snd_printk(KERN_ERR "unable to find a free DMA2\n"); - return -EBUSY; - } - } + /* init proc interface */ + snd_miro_proc_init(card, miro); error = snd_miro_configure(miro); - if (error) { - snd_card_free(card); + if (error) return error; - } error = snd_wss_create(card, miro->wss_base + 4, -1, - miro->irq, miro->dma1, miro->dma2, - WSS_HW_AD1845, 0, &codec); - if (error < 0) { - snd_card_free(card); + miro->irq, miro->dma1, miro->dma2, + WSS_HW_DETECT, 0, &codec); + if (error < 0) return error; - } error = snd_wss_pcm(codec, 0, &pcm); - if (error < 0) { - snd_card_free(card); + if (error < 0) return error; - } + error = snd_wss_mixer(codec); - if (error < 0) { - snd_card_free(card); + if (error < 0) return error; - } + error = snd_wss_timer(codec, 0, &timer); - if (error < 0) { - snd_card_free(card); + if (error < 0) return error; - } miro->pcm = pcm; error = snd_miro_mixer(card, miro); - if (error < 0) { - snd_card_free(card); + if (error < 0) return error; - } if (miro->aci->aci_vendor == 'm') { /* It looks like a miro sound card. */ @@ -1425,20 +1354,111 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) if (fm_port > 0 && fm_port != SNDRV_AUTO_PORT) { struct snd_opl3 *opl3 = NULL; struct snd_opl4 *opl4; + if (snd_opl4_create(card, fm_port, fm_port - 8, 2, &opl3, &opl4) < 0) snd_printk(KERN_WARNING "no OPL4 device at 0x%lx\n", fm_port); } - if ((error = snd_set_aci_init_values(miro)) < 0) { - snd_card_free(card); + error = snd_set_aci_init_values(miro); + if (error < 0) return error; + + return snd_card_register(card); +} + +static int __devinit snd_miro_isa_match(struct device *devptr, unsigned int n) +{ + return 1; +} + +static int __devinit snd_miro_isa_probe(struct device *devptr, unsigned int n) +{ + static long possible_ports[] = {0x530, 0xe80, 0xf40, 0x604, -1}; + static long possible_mpu_ports[] = {0x330, 0x300, 0x310, 0x320, -1}; + static int possible_irqs[] = {11, 9, 10, 7, -1}; + static int possible_mpu_irqs[] = {10, 5, 9, 7, -1}; + static int possible_dma1s[] = {3, 1, 0, -1}; + static int possible_dma2s[][2] = { {1, -1}, {0, -1}, {-1, -1}, + {0, -1} }; + + int error; + struct snd_miro *miro; + struct snd_card *card; + + error = snd_card_create(index, id, THIS_MODULE, + sizeof(struct snd_miro), &card); + if (error < 0) + return error; + + card->private_free = snd_card_miro_free; + miro = card->private_data; + + error = snd_card_miro_detect(card, miro); + if (error < 0) { + snd_card_free(card); + snd_printk(KERN_ERR "unable to detect OPTi9xx chip\n"); + return -ENODEV; + } + + if (port == SNDRV_AUTO_PORT) { + port = snd_legacy_find_free_ioport(possible_ports, 4); + if (port < 0) { + snd_card_free(card); + snd_printk(KERN_ERR "unable to find a free WSS port\n"); + return -EBUSY; + } + } + + if (mpu_port == SNDRV_AUTO_PORT) { + mpu_port = snd_legacy_find_free_ioport(possible_mpu_ports, 2); + if (mpu_port < 0) { + snd_card_free(card); + snd_printk(KERN_ERR + "unable to find a free MPU401 port\n"); + return -EBUSY; + } + } + + if (irq == SNDRV_AUTO_IRQ) { + irq = snd_legacy_find_free_irq(possible_irqs); + if (irq < 0) { + snd_card_free(card); + snd_printk(KERN_ERR "unable to find a free IRQ\n"); + return -EBUSY; + } + } + if (mpu_irq == SNDRV_AUTO_IRQ) { + mpu_irq = snd_legacy_find_free_irq(possible_mpu_irqs); + if (mpu_irq < 0) { + snd_card_free(card); + snd_printk(KERN_ERR + "unable to find a free MPU401 IRQ\n"); + return -EBUSY; + } + } + if (dma1 == SNDRV_AUTO_DMA) { + dma1 = snd_legacy_find_free_dma(possible_dma1s); + if (dma1 < 0) { + snd_card_free(card); + snd_printk(KERN_ERR "unable to find a free DMA1\n"); + return -EBUSY; + } + } + if (dma2 == SNDRV_AUTO_DMA) { + dma2 = snd_legacy_find_free_dma(possible_dma2s[dma1 % 4]); + if (dma2 < 0) { + snd_card_free(card); + snd_printk(KERN_ERR "unable to find a free DMA2\n"); + return -EBUSY; + } } snd_card_set_dev(card, devptr); - if ((error = snd_card_register(card))) { + error = snd_miro_probe(card); + if (error < 0) { snd_card_free(card); return error; } @@ -1447,7 +1467,8 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) return 0; } -static int __devexit snd_miro_remove(struct device *devptr, unsigned int dev) +static int __devexit snd_miro_isa_remove(struct device *devptr, + unsigned int dev) { snd_card_free(dev_get_drvdata(devptr)); dev_set_drvdata(devptr, NULL); @@ -1457,9 +1478,9 @@ static int __devexit snd_miro_remove(struct device *devptr, unsigned int dev) #define DEV_NAME "miro" static struct isa_driver snd_miro_driver = { - .match = snd_miro_match, - .probe = snd_miro_probe, - .remove = __devexit_p(snd_miro_remove), + .match = snd_miro_isa_match, + .probe = snd_miro_isa_probe, + .remove = __devexit_p(snd_miro_isa_remove), /* FIXME: suspend/resume */ .driver = { .name = DEV_NAME -- 1.6.4 ---------------------------------------------------------------------- Wygraj pobyt w Alpach dla ca�ej rodziny Kliknij >>> http://link.interia.pl/f2446

2 1

[alsa-devel] [PATCH 2/2] ALSA: AACI: fix recording bug
by Russell King - ARM Linux 30 Nov '09

30 Nov '09

pcm->r[1].slots is the double rate slot information, not the capture information. For capture, 'pcm' will already be the capture ac97 pcm structure. Signed-off-by: Russell King <rmk+kernel(a)arm.linux.org.uk> --- sound/arm/aaci.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 1456fc7..eb715e7 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -518,7 +518,7 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream, else err = snd_ac97_pcm_open(aacirun->pcm, params_rate(params), params_channels(params), - aacirun->pcm->r[1].slots); + aacirun->pcm->r[0].slots); if (err) goto out; -- 1.6.2.5

2 3

[alsa-devel] [PATCH 2/2] opti93x: set MC indirect registers base from PnP data
by Krzysztof Helt 30 Nov '09

30 Nov '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The PnP data on the OPTI931 and OPTI933 contains io port range for the MC indirect registers. Use the PnP range instead of hardwired value 0xE0E. Also, request region of MC indirect registers so it is marked as used to other drivers (this was missing previously). Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/opti9xx/opti92x-ad1848.c | 53 +++++++++++++++++++++++++++++------ 1 files changed, 44 insertions(+), 9 deletions(-) diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 438c1da..07ab288 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -135,6 +135,8 @@ struct snd_opti9xx { unsigned long mc_base_size; #ifdef OPTi93X unsigned long mc_indir_index; + unsigned long mc_indir_size; + struct resource *res_mc_indir; struct snd_wss *codec; #endif /* OPTi93X */ unsigned long pwd_reg; @@ -245,7 +247,10 @@ static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip, case OPTi9XX_HW_82C931: case OPTi9XX_HW_82C933: chip->mc_base = (hardware == OPTi9XX_HW_82C930) ? 0xf8f : 0xf8d; - chip->mc_indir_index = 0xe0e; + if (!chip->mc_indir_index) { + chip->mc_indir_index = 0xe0e; + chip->mc_indir_size = 2; + } chip->password = 0xe4; chip->pwd_reg = 0; break; @@ -579,10 +584,10 @@ static int __devinit snd_card_opti9xx_detect(struct snd_card *card, struct snd_opti9xx *chip) { int i, err; + unsigned char value; #ifndef OPTi93X for (i = OPTi9XX_HW_82C928; i < OPTi9XX_HW_82C930; i++) { - unsigned char value; if ((err = snd_opti9xx_init(chip, i)) < 0) return err; @@ -602,18 +607,26 @@ static int __devinit snd_card_opti9xx_detect(struct snd_card *card, #else /* OPTi93X */ for (i = OPTi9XX_HW_82C931; i >= OPTi9XX_HW_82C930; i--) { unsigned long flags; - unsigned char value; - if ((err = snd_opti9xx_init(chip, i)) < 0) + err = snd_opti9xx_init(chip, i); + if (err < 0) return err; - if ((chip->res_mc_base = request_region(chip->mc_base, chip->mc_base_size, "OPTi9xx MC")) == NULL) - continue; + chip->res_mc_base = request_region(chip->mc_base, + chip->mc_base_size, + "OPTi9xx MC"); + if (chip->res_mc_base == NULL) + return -EBUSY; + + chip->res_mc_indir = request_region(chip->mc_indir_index, + chip->mc_indir_size, + "OPTi93x MC"); + if (chip->res_mc_indir == NULL) + return -EBUSY; spin_lock_irqsave(&chip->lock, flags); outb(chip->password, chip->mc_base + chip->pwd_reg); - outb(((chip->mc_indir_index & (1 << 8)) >> 4) | - ((chip->mc_indir_index & 0xf0) >> 4), chip->mc_base); + outb(((chip->mc_indir_index & 0x1f0) >> 4), chip->mc_base); spin_unlock_irqrestore(&chip->lock, flags); value = snd_opti9xx_read(chip, OPTi9XX_MC_REG(7)); @@ -621,6 +634,8 @@ static int __devinit snd_card_opti9xx_detect(struct snd_card *card, if (snd_opti9xx_read(chip, OPTi9XX_MC_REG(7)) == 0xff - value) return 1; + release_and_free_resource(chip->res_mc_indir); + chip->res_mc_indir = NULL; release_and_free_resource(chip->res_mc_base); chip->res_mc_base = NULL; } @@ -654,6 +669,8 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip, #ifdef OPTi93X port = pnp_port_start(pdev, 0) - 4; fm_port = pnp_port_start(pdev, 1) + 8; + chip->mc_indir_index = pnp_port_start(pdev, 3) + 2; + chip->mc_indir_size = pnp_port_len(pdev, 3) - 2; #else if (pid->driver_data != 0x0924) port = pnp_port_start(pdev, 1); @@ -684,7 +701,7 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip, static void snd_card_opti9xx_free(struct snd_card *card) { struct snd_opti9xx *chip = card->private_data; - + if (chip) { #ifdef OPTi93X struct snd_wss *codec = chip->codec; @@ -692,6 +709,7 @@ static void snd_card_opti9xx_free(struct snd_card *card) disable_irq(chip->irq); free_irq(chip->irq, codec); } + release_and_free_resource(chip->res_mc_indir); #endif release_and_free_resource(chip->res_mc_base); } @@ -709,12 +727,29 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) struct snd_pcm *pcm; struct snd_rawmidi *rmidi; struct snd_hwdep *synth; +#ifdef OPTi93X + unsigned long flags; +#endif if (! chip->res_mc_base && (chip->res_mc_base = request_region(chip->mc_base, chip->mc_base_size, "OPTi9xx MC")) == NULL) return -ENOMEM; +#ifdef OPTi93X + if (chip->res_mc_indir == NULL) { + chip->res_mc_indir = request_region(chip->mc_indir_index, + chip->mc_indir_size, + "OPTi93x MC"); + if (chip->res_mc_indir == NULL) + return -ENOMEM; + } + /* store MC indirect range address */ + spin_lock_irqsave(&chip->lock, flags); + outb(chip->password, chip->mc_base + chip->pwd_reg); + outb(((chip->mc_indir_index & 0x1f0) >> 4), chip->mc_base); + spin_unlock_irqrestore(&chip->lock, flags); +#endif chip->wss_base = port; chip->fm_port = fm_port; chip->mpu_port = mpu_port; -- 1.6.4 ---------------------------------------------------------------------- Codziennie 11 tys. ofert pracy! Sprawdz: http://link.interia.pl/f2447

2 1

[alsa-devel] [PATCH] ALSA: ASoc: Add regulator support to CS4270 codec driver
by Daniel Mack 30 Nov '09

30 Nov '09

Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Timur Tabi <timur(a)freescale.com> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> --- sound/soc/codecs/cs4270.c | 35 +++++++++++++++++++++++++++++++---- 1 files changed, 31 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 8069023..b1d4b5d 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -28,6 +28,7 @@ #include <sound/initval.h> #include <linux/i2c.h> #include <linux/delay.h> +#include <linux/regulator/consumer.h> #include "cs4270.h" @@ -114,6 +115,7 @@ struct cs4270_private { unsigned int mode; /* The mode (I2S or left-justified) */ unsigned int slave_mode; unsigned int manual_mute; + struct regulator *va_reg; }; /** @@ -462,27 +464,38 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, * @dai: the SOC DAI * @mute: 0 = disable mute, 1 = enable mute * - * This function toggles the mute bits in the MUTE register. The CS4270's + * This function toggles the mute bits in the MUTE register and switches + * the registered VA power regulator, if it was registered. The CS4270's * mute capability is intended for external muting circuitry, so if the * board does not have the MUTEA or MUTEB pins connected to such circuitry, - * then this function will do nothing. + * and no power regulator was specified, then this function will do nothing. */ static int cs4270_dai_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; struct cs4270_private *cs4270 = codec->private_data; - int reg6; + int reg6, err; reg6 = snd_soc_read(codec, CS4270_MUTE); if (mute) reg6 |= CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B; else { + if (cs4270->va_reg) + regulator_enable(cs4270->va_reg); + reg6 &= ~(CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B); reg6 |= cs4270->manual_mute; } - return snd_soc_write(codec, CS4270_MUTE, reg6); + err = snd_soc_write(codec, CS4270_MUTE, reg6); + if (err) + return err; + + if (cs4270->va_reg && mute) + regulator_disable(cs4270->va_reg); + + return 0; } /** @@ -579,6 +592,7 @@ static int cs4270_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = cs4270_codec; + struct cs4270_private *cs4270 = codec->private_data; int ret; /* Connect the codec to the socdev. snd_soc_new_pcms() needs this. */ @@ -606,6 +620,14 @@ static int cs4270_probe(struct platform_device *pdev) goto error_free_pcms; } +#ifdef CONFIG_REGULATOR + /* get the regulator if there is one read<y for us */ + cs4270->va_reg = regulator_get(&pdev->dev, "va"); + + if (IS_ERR(cs4270->va_reg)) + cs4270->va_reg = NULL; +#endif + return 0; error_free_pcms: @@ -623,9 +645,14 @@ error_free_pcms: static int cs4270_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = cs4270_codec; + struct cs4270_private *cs4270 = codec->private_data; snd_soc_free_pcms(socdev); + if (cs4270->va_reg) + regulator_put(cs4270->va_reg); + return 0; }; -- 1.6.5.2

3 17

[alsa-devel] [PATCH 1/2] opti-93x: fix irq freeing after request_irq error
by Krzysztof Helt 30 Nov '09

30 Nov '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Set snd_opti->irq value to -1 if the request_irq() fails to avoid freeing the irq inside the snd_card_free(). Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/opti9xx/opti92x-ad1848.c | 7 ++++--- 1 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 5cd5553..438c1da 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -688,9 +688,9 @@ static void snd_card_opti9xx_free(struct snd_card *card) if (chip) { #ifdef OPTi93X struct snd_wss *codec = chip->codec; - if (codec && codec->irq > 0) { - disable_irq(codec->irq); - free_irq(codec->irq, codec); + if (codec && chip->irq > 0) { + disable_irq(chip->irq); + free_irq(chip->irq, codec); } #endif release_and_free_resource(chip->res_mc_base); @@ -767,6 +767,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) IRQF_DISABLED, DEV_NAME" - WSS", codec); if (error < 0) { snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", chip->irq); + chip->irq = -1; return error; } #endif -- 1.6.4 ---------------------------------------------------------------------- Codziennie 11 tys. ofert pracy! Sprawdz: http://link.interia.pl/f2447

2 1

[PATCH] ALSA: ASoc: pxa/raumfeld: adopt new snd_soc_dai_set_pll() API
by Daniel Mack 30 Nov '09

30 Nov '09

ALSA's for-2.6.33 branch has a new source argument to snd_soc_dai_set_pll(). Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/pxa/raumfeld.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 5d7bb8c..e713181 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -116,7 +116,7 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, return ret; /* setup the CPU DAI */ - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, clk); + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk); if (ret < 0) return ret; @@ -205,7 +205,7 @@ static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream, return ret; /* setup the CPU DAI */ - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, clk); + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk); if (ret < 0) return ret; -- 1.6.5.2

1 0

[alsa-devel] [PATCH 2/2] opti-miro: add PnP detection
by Krzysztof Helt 30 Nov '09

30 Nov '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The PCM12 and PCM20 can be set into the ISA PnP mode. The PCM12 PnP was sold as the PnP device. Add code to handle detection of these cards using ISA PnP framework. Tested on the PCM20 in PnP mode. The PCM12 PnP has the same MS Windows INF file except for a card name displayed for user. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/opti9xx/miro.c | 203 +++++++++++++++++++++++++++++++++++++++++++--- 1 files changed, 192 insertions(+), 11 deletions(-) diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index c67bc3c..6123c75 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -25,6 +25,7 @@ #include <linux/init.h> #include <linux/err.h> #include <linux/isa.h> +#include <linux/pnp.h> #include <linux/delay.h> #include <linux/slab.h> #include <linux/ioport.h> @@ -60,6 +61,9 @@ static int dma1 = SNDRV_DEFAULT_DMA1; /* 0,1,3 */ static int dma2 = SNDRV_DEFAULT_DMA1; /* 0,1,3 */ static int wss; static int ide; +#ifdef CONFIG_PNP +static int isapnp = 1; /* Enable ISA PnP detection */ +#endif module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for miro soundcard."); @@ -83,6 +87,10 @@ module_param(wss, int, 0444); MODULE_PARM_DESC(wss, "wss mode"); module_param(ide, int, 0444); MODULE_PARM_DESC(ide, "enable ide port"); +#ifdef CONFIG_PNP +module_param(isapnp, bool, 0444); +MODULE_PARM_DESC(isapnp, "Enable ISA PnP detection for specified soundcard."); +#endif #define OPTi9XX_HW_DETECT 0 #define OPTi9XX_HW_82C928 1 @@ -131,6 +139,21 @@ static char * snd_opti9xx_names[] = { "82C930", "82C931", "82C933" }; +static int snd_miro_pnp_is_probed; + +#ifdef CONFIG_PNP + +static struct pnp_card_device_id snd_miro_pnpids[] = { + /* PCM20 and PCM12 in PnP mode */ + { .id = "MIR0924", + .devs = { { "MIR0000" }, { "MIR0002" }, { "MIR0005" } }, }, + { .id = "" } +}; + +MODULE_DEVICE_TABLE(pnp_card, snd_miro_pnpids); + +#endif /* CONFIG_PNP */ + /* * ACI control */ @@ -781,17 +804,23 @@ static int __devinit snd_miro_init(struct snd_miro *chip, chip->mpu_port = -1; chip->mpu_irq = -1; + chip->pwd_reg = 3; + +#ifdef CONFIG_PNP + if (isapnp && chip->mc_base) + /* PnP resource gives the least 10 bits */ + chip->mc_base |= 0xc00; + else +#endif + chip->mc_base = 0xf8c; + switch (hardware) { case OPTi9XX_HW_82C929: - chip->mc_base = 0xf8c; chip->password = 0xe3; - chip->pwd_reg = 3; break; case OPTi9XX_HW_82C924: - chip->mc_base = 0xf8c; chip->password = 0xe5; - chip->pwd_reg = 3; break; default: @@ -1014,17 +1043,22 @@ static int __devinit snd_miro_configure(struct snd_miro *chip) return -EINVAL; } - switch (chip->wss_base) { - case 0x530: + /* PnP resource says it decodes only 10 bits of address */ + switch (chip->wss_base & 0x3ff) { + case 0x130: + chip->wss_base = 0x530; wss_base_bits = 0x00; break; - case 0x604: + case 0x204: + chip->wss_base = 0x604; wss_base_bits = 0x03; break; - case 0xe80: + case 0x280: + chip->wss_base = 0xe80; wss_base_bits = 0x01; break; - case 0xf40: + case 0x340: + chip->wss_base = 0xf40; wss_base_bits = 0x02; break; default: @@ -1238,7 +1272,7 @@ static int __devinit snd_card_miro_aci_detect(struct snd_card *card, static void snd_card_miro_free(struct snd_card *card) { struct snd_miro *miro = card->private_data; - + release_and_free_resource(miro->res_aci_port); if (miro->aci) miro->aci->aci_port = 0; @@ -1370,6 +1404,12 @@ static int __devinit snd_miro_probe(struct snd_card *card) static int __devinit snd_miro_isa_match(struct device *devptr, unsigned int n) { +#ifdef CONFIG_PNP + if (snd_miro_pnp_is_probed) + return 0; + if (isapnp) + return 0; +#endif return 1; } @@ -1487,14 +1527,155 @@ static struct isa_driver snd_miro_driver = { }, }; +#ifdef CONFIG_PNP + +static int __devinit snd_card_miro_pnp(struct snd_miro *chip, + struct pnp_card_link *card, + const struct pnp_card_device_id *pid) +{ + struct pnp_dev *pdev; + int err; + struct pnp_dev *devmpu; + struct pnp_dev *devmc; + + pdev = pnp_request_card_device(card, pid->devs[0].id, NULL); + if (pdev == NULL) + return -EBUSY; + + devmpu = pnp_request_card_device(card, pid->devs[1].id, NULL); + if (devmpu == NULL) + return -EBUSY; + + devmc = pnp_request_card_device(card, pid->devs[2].id, NULL); + if (devmc == NULL) + return -EBUSY; + + err = pnp_activate_dev(pdev); + if (err < 0) { + snd_printk(KERN_ERR "AUDIO pnp configure failure: %d\n", err); + return err; + } + + err = pnp_activate_dev(devmc); + if (err < 0) { + snd_printk(KERN_ERR "OPL syntg pnp configure failure: %d\n", + err); + return err; + } + + port = pnp_port_start(pdev, 1); + fm_port = pnp_port_start(pdev, 2) + 8; + + /* + * The MC(0) is never accessed and the miroSOUND PCM20 card does not + * include it in the PnP resource range. OPTI93x include it. + */ + chip->mc_base = pnp_port_start(devmc, 0) - 1; + chip->mc_base_size = pnp_port_len(devmc, 0) + 1; + + irq = pnp_irq(pdev, 0); + dma1 = pnp_dma(pdev, 0); + dma2 = pnp_dma(pdev, 1); + + if (mpu_port > 0) { + err = pnp_activate_dev(devmpu); + if (err < 0) { + snd_printk(KERN_ERR "MPU401 pnp configure failure\n"); + mpu_port = -1; + return err; + } + mpu_port = pnp_port_start(devmpu, 0); + mpu_irq = pnp_irq(devmpu, 0); + } + return 0; +} + +static int __devinit snd_miro_pnp_probe(struct pnp_card_link *pcard, + const struct pnp_card_device_id *pid) +{ + struct snd_card *card; + int err; + struct snd_miro *miro; + + if (snd_miro_pnp_is_probed) + return -EBUSY; + if (!isapnp) + return -ENODEV; + err = snd_card_create(index, id, THIS_MODULE, + sizeof(struct snd_miro), &card); + if (err < 0) + return err; + + card->private_free = snd_card_miro_free; + miro = card->private_data; + + err = snd_card_miro_pnp(miro, pcard, pid); + if (err) { + snd_card_free(card); + return err; + } + + /* only miroSOUND PCM20 and PCM12 == OPTi924 */ + err = snd_miro_init(miro, OPTi9XX_HW_82C924); + if (err) { + snd_card_free(card); + return err; + } + + err = snd_miro_opti_check(miro); + if (err) { + snd_printk(KERN_ERR "OPTI chip not found\n"); + snd_card_free(card); + return err; + } + + snd_card_set_dev(card, &pcard->card->dev); + err = snd_miro_probe(card); + if (err < 0) { + snd_card_free(card); + return err; + } + pnp_set_card_drvdata(pcard, card); + snd_miro_pnp_is_probed = 1; + return 0; +} + +static void __devexit snd_miro_pnp_remove(struct pnp_card_link * pcard) +{ + snd_card_free(pnp_get_card_drvdata(pcard)); + pnp_set_card_drvdata(pcard, NULL); + snd_miro_pnp_is_probed = 0; +} + +static struct pnp_card_driver miro_pnpc_driver = { + .flags = PNP_DRIVER_RES_DISABLE, + .name = "miro", + .id_table = snd_miro_pnpids, + .probe = snd_miro_pnp_probe, + .remove = __devexit_p(snd_miro_pnp_remove), +}; +#endif + static int __init alsa_card_miro_init(void) { +#ifdef CONFIG_PNP + pnp_register_card_driver(&miro_pnpc_driver); + if (snd_miro_pnp_is_probed) + return 0; + pnp_unregister_card_driver(&miro_pnpc_driver); +#endif return isa_register_driver(&snd_miro_driver, 1); } static void __exit alsa_card_miro_exit(void) { - isa_unregister_driver(&snd_miro_driver); + if (!snd_miro_pnp_is_probed) { + isa_unregister_driver(&snd_miro_driver); + return; + } +#ifdef CONFIG_PNP + pnp_unregister_card_driver(&miro_pnpc_driver); +#endif } module_init(alsa_card_miro_init) -- 1.6.4 ---------------------------------------------------------------------- Konkurs dla fanow Hannah Montana! 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1 0

[alsa-devel] [PATCH 1/2] ALSA: AACI: fix AC97 multiple-open bug
by Russell King - ARM Linux 29 Nov '09

29 Nov '09

Signed-off-by: Russell King <rmk+kernel(a)arm.linux.org.uk> --- sound/arm/aaci.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index a03fe80..1456fc7 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -501,6 +501,10 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream, int err; aaci_pcm_hw_free(substream); + if (aacirun->pcm_open) { + snd_ac97_pcm_close(aacirun->pcm); + aacirun->pcm_open = 0; + } err = devdma_hw_alloc(NULL, substream, params_buffer_bytes(params)); -- 1.6.2.5

1 0

[alsa-devel] i.MX audio support
by Sascha Hauer 29 Nov '09

29 Nov '09

Hi All, This series adds support for AOSC on i.MX platforms. It uses DMA where available (i.MX27) and fiq otherwise. The driver works on all Phytec i.MX platforms, so the i.MX27, i.MX31 and the i.MX35 are supported including AC97 support. There already is a driver for i.MX in the tree, this one is a nearly complete rewrite. I haven't touched the in Kernel driver with these patches. The in Kernel driver has several problems, it doesn't use ioremap, uses direct pointer derefs instead of proper access functions and the only board supported so far is itself not supported in mainline. AC97 support is a problem on i.MX. I found no way to tell the SSI controller which slots it should capture, so all slots the codec decides to send go into one single FIFO. The WM9712 I worked with sends data in slot 12 which we have to sort out manually from the FIFO, so I think there is no way to work around the fiq handler using the DMA engine with AC97. The devdma code is actually a copy of sound/arm/devdma.[ch]. It should probably be selected somehow instead of duplicated. I'm not very familiar with ALSA, so I expect some comments in this area. Let me know what has to be done to get this into mainline. This series depends on the SSI resources and audmux support being present. I posted these patches on arm linux kernel recently and they are available here: git://git.pengutronix.de/git/imx/linux-2.6.git mxc-master Any comments welcome. Sascha The following changes since commit 3f92a8bd5fb13e7e2505c65d1548910eaa843024: Uwe Kleine-König (1): imx: copy constants from mx3x.h to mx35.h using the appropriate namespace are available in the git repository at: git://git.pengutronix.de/git/imx/linux-2.6.git asoc Sascha Hauer (12): mx3: Add SSI pins to iomux table mxc: iomux v3: remove resource handling add a mc13783 codec driver i.MX31 clock: rename SSI clocks to driver name mxc: mx1/mx2 DMA: add a possibility to create an endless DMA transfer imx-ssi sound driver add phycore sound support sound/soc/imx: Makefile/Kconfig changes for new driver pcm038: add sound support pcm043: add sound support pca100: add sound support pcm037: Add sound support arch/arm/mach-mx2/pca100.c | 52 + arch/arm/mach-mx2/pcm038.c | 23 +- arch/arm/mach-mx3/clock.c | 4 +- arch/arm/mach-mx3/pcm037.c | 72 ++ arch/arm/mach-mx3/pcm043.c | 102 ++ arch/arm/plat-mxc/Makefile | 1 + arch/arm/plat-mxc/dma-mx1-mx2.c | 3 +- arch/arm/plat-mxc/include/mach/dma-mx1-mx2.h | 8 + arch/arm/plat-mxc/include/mach/iomux-mx3.h | 8 + arch/arm/plat-mxc/include/mach/iomux-v3.h | 17 +- arch/arm/plat-mxc/include/mach/ssi.h | 17 + arch/arm/plat-mxc/iomux-v3.c | 36 +- arch/arm/plat-mxc/ssi-fiq.S | 134 +++ sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mc13783.c | 739 ++++++++++++++ sound/soc/codecs/mc13783.h | 29 + sound/soc/imx/Kconfig | 15 +- sound/soc/imx/Makefile | 9 +- sound/soc/imx/devdma.c | 80 ++ sound/soc/imx/devdma.h | 3 + sound/soc/imx/imx-ssi.c | 1349 ++++++++++++++++++++++++++ sound/soc/imx/imx-ssi.h | 24 + sound/soc/imx/phycore.c | 181 ++++ 24 files changed, 2851 insertions(+), 60 deletions(-) create mode 100644 arch/arm/plat-mxc/include/mach/ssi.h create mode 100644 arch/arm/plat-mxc/ssi-fiq.S create mode 100644 sound/soc/codecs/mc13783.c create mode 100644 sound/soc/codecs/mc13783.h create mode 100644 sound/soc/imx/devdma.c create mode 100644 sound/soc/imx/devdma.h create mode 100644 sound/soc/imx/imx-ssi.c create mode 100644 sound/soc/imx/imx-ssi.h create mode 100644 sound/soc/imx/phycore.c

4 39

Re: [alsa-devel] [Alsa-user] m-audio delta 1010 inputs crackle at 96 kHz sample rate with all newer kernels
by Clemens Ladisch 27 Nov '09

27 Nov '09

francis keyes wrote: > As soon as I switch to 96kHz in envy24control, I can hear lots of > crackle and pop noises through the inputs that are also recorded when > I try to record. I first noticed this after installing a new kernel. Which kernel? Best regards, Clemens

2 1

[alsa-devel] [PATCH] ASoC: Implement initial S3C64XX IISv4 support
by Mark Brown 27 Nov '09

27 Nov '09

The IISv4 port can be controlled by the IISv2 driver so copy over that with light modifications, including use of platform data to get the pin configuration. Currently this is S3C6410 specific due to the DMA configuration - in the future when we have IISv4 specific support we may want to split out a generic IISv4 portion. Since this allows the SMDK64xx machine driver to work update it to reflect the API changes involved and mark it as not BROKEN. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- This depends on the arch/arm changes I posted to linux-arm-kernel a minute ago. I'll not push it until it looks like those will be merged. sound/soc/s3c24xx/Kconfig | 10 +++++++--- sound/soc/s3c24xx/Makefile | 2 ++ sound/soc/s3c24xx/s3c64xx-i2s.h | 1 + sound/soc/s3c24xx/smdk64xx_wm8580.c | 6 ++---- 4 files changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index b489f1a..1b98850 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 @@ -62,11 +67,10 @@ config SND_S3C24XX_SOC_JIVE_WM8750 config SND_S3C64XX_SOC_WM8580 tristate "SoC I2S Audio support for WM8580 on SMDK64XX" depends on SND_S3C24XX_SOC && (MACH_SMDK6400 || MACH_SMDK6410) - depends on BROKEN select SND_SOC_WM8580 - select SND_S3C64XX_SOC_I2S + select SND_S3C64XX_SOC_I2S_V4 help - Sat Y if you want to add support for SoC audio on the SMDK64XX. + Say Y if you want to add support for SoC audio on the SMDK64XX. config SND_S3C24XX_SOC_SMDK2443_WM9710 tristate "SoC AC97 Audio support for SMDK2443 - WM9710" diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index b744657..114028c 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -3,6 +3,7 @@ snd-soc-s3c24xx-objs := s3c-dma.o 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-s3c64xx-i2s-v4-objs := s3c64xx-i2s-v4.o snd-soc-s3c2443-ac97-objs := s3c2443-ac97.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_S3C2443_SOC_AC97) += snd-soc-s3c2443-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.h b/sound/soc/s3c24xx/s3c64xx-i2s.h index abe7253..a50b2dc 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.h +++ b/sound/soc/s3c24xx/s3c64xx-i2s.h @@ -28,6 +28,7 @@ struct clk; #define S3C64XX_CLKSRC_CDCLK (2) extern struct snd_soc_dai s3c64xx_i2s_dai[]; +extern struct snd_soc_dai s3c64xx_i2s_v4_dai[]; extern struct clk *s3c64xx_i2s_get_clock(struct snd_soc_dai *dai); diff --git a/sound/soc/s3c24xx/smdk64xx_wm8580.c b/sound/soc/s3c24xx/smdk64xx_wm8580.c index efe4901..b5bb129 100644 --- a/sound/soc/s3c24xx/smdk64xx_wm8580.c +++ b/sound/soc/s3c24xx/smdk64xx_wm8580.c @@ -22,8 +22,6 @@ #include "s3c-dma.h" #include "s3c64xx-i2s.h" -#define S3C64XX_I2S_V4 2 - /* SMDK64XX has a 12MHZ crystal attached to WM8580 */ #define SMDK64XX_WM8580_FREQ 12000000 @@ -215,7 +213,7 @@ static struct snd_soc_dai_link smdk64xx_dai[] = { { /* Primary Playback i/f */ .name = "WM8580 PAIF RX", .stream_name = "Playback", - .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4], + .cpu_dai = &s3c64xx_i2s_v4_dai[0], .codec_dai = &wm8580_dai[WM8580_DAI_PAIFRX], .init = smdk64xx_wm8580_init_paifrx, .ops = &smdk64xx_ops, @@ -223,7 +221,7 @@ static struct snd_soc_dai_link smdk64xx_dai[] = { { /* Primary Capture i/f */ .name = "WM8580 PAIF TX", .stream_name = "Capture", - .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4], + .cpu_dai = &s3c64xx_i2s_v4_dai[0], .codec_dai = &wm8580_dai[WM8580_DAI_PAIFTX], .init = smdk64xx_wm8580_init_paiftx, .ops = &smdk64xx_ops, -- 1.6.5.3

1 0

[alsa-devel] [PATCH 2/3] ALSA: ARM: add Raumfeld audio support
by Daniel Mack 27 Nov '09

27 Nov '09

Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/pxa/Kconfig | 9 ++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/raumfeld.c | 335 ++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 346 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/raumfeld.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index dcb3181..9c3bfd9 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -117,6 +117,15 @@ config SND_SOC_ZYLONITE Say Y if you want to add support for SoC audio on the Marvell Zylonite reference platform. +config SND_SOC_RAUMFELD + tristate "SoC Audio support Raumfeld audio adapter" + depends on SND_PXA2XX_SOC && (MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR) + select SND_PXA_SOC_SSP + select SND_SOC_CS4270 + select SND_SOC_AK4104 + help + Say Y if you want to add support for SoC audio on Raumfeld devices + config SND_PXA2XX_SOC_MAGICIAN tristate "SoC Audio support for HTC Magician" depends on SND_PXA2XX_SOC && MACH_MAGICIAN diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 6e096b4..f3e08fd 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -23,6 +23,7 @@ snd-soc-zylonite-objs := zylonite.o snd-soc-magician-objs := magician.o snd-soc-mioa701-objs := mioa701_wm9713.o snd-soc-imote2-objs := imote2.o +snd-soc-raumfeld-objs := raumfeld.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o @@ -37,3 +38,4 @@ obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o +obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c new file mode 100644 index 0000000..f272269 --- /dev/null +++ b/sound/soc/pxa/raumfeld.c @@ -0,0 +1,335 @@ +/* + * raumfeld_audio.c -- SoC audio for Raumfeld audio devices + * + * Copyright (c) 2009 Daniel Mack <daniel(a)caiaq.de> + * + * based on code from: + * + * Wolfson Microelectronics PLC. + * Openedhand Ltd. + * Liam Girdwood <lrg(a)slimlogic.co.uk> + * Richard Purdie <richard(a)openedhand.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; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/module.h> +#include <linux/i2c.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> + +#include "../codecs/cs4270.h" +#include "../codecs/ak4104.h" +#include "pxa2xx-pcm.h" +#include "pxa-ssp.h" + +#define GPIO_SPDIF_RESET (38) +#define GPIO_MCLK_RESET (111) +#define GPIO_CODEC_RESET (120) + +static struct i2c_client *max9486_client; +static struct i2c_board_info max9486_hwmon_info = { + I2C_BOARD_INFO("max9485", 0x63), +}; + +#define MAX9485_MCLK_FREQ_112896 0x22 +#define MAX9485_MCLK_FREQ_122880 0x23 + +static void set_max9485_clk(char clk) +{ + i2c_master_send(max9486_client, &clk, 1); +} + +static void raumfeld_enable_audio(bool en) +{ + if (en) { + gpio_set_value(GPIO_MCLK_RESET, 1); + + /* wait some time to let the clocks become stable */ + msleep(100); + + gpio_set_value(GPIO_SPDIF_RESET, 1); + gpio_set_value(GPIO_CODEC_RESET, 1); + } else { + gpio_set_value(GPIO_MCLK_RESET, 0); + gpio_set_value(GPIO_SPDIF_RESET, 0); + gpio_set_value(GPIO_CODEC_RESET, 0); + } +} + +/* CS4270 */ +static int raumfeld_cs4270_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + set_max9485_clk(MAX9485_MCLK_FREQ_112896); + + return snd_soc_dai_set_sysclk(codec_dai, 0, 11289600, 0); +} + +static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int fmt, clk = 0; + int ret = 0; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + set_max9485_clk(MAX9485_MCLK_FREQ_122880); + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + set_max9485_clk(MAX9485_MCLK_FREQ_112896); + clk = 11289600; + break; + } + + fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; + + /* setup the CODEC DAI */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, 0); + if (ret < 0) + return ret; + + /* setup the CPU DAI */ + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, clk); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, 0, 1); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops raumfeld_cs4270_ops = { + .startup = raumfeld_cs4270_startup, + .hw_params = raumfeld_cs4270_hw_params, +}; + +static int raumfeld_line_suspend(struct platform_device *pdev, pm_message_t state) +{ + raumfeld_enable_audio(false); + return 0; +} + +static int raumfeld_line_resume(struct platform_device *pdev) +{ + raumfeld_enable_audio(true); + return 0; +} + +static struct snd_soc_dai_link raumfeld_line_dai = { + .name = "CS4270", + .stream_name = "CS4270", + .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP1], + .codec_dai = &cs4270_dai, + .ops = &raumfeld_cs4270_ops, +}; + +static struct snd_soc_card snd_soc_line_raumfeld = { + .name = "Raumfeld analog", + .platform = &pxa2xx_soc_platform, + .dai_link = &raumfeld_line_dai, + .suspend_post = raumfeld_line_suspend, + .resume_pre = raumfeld_line_resume, + .num_links = 1, +}; + + +/* AK4104 */ + +static int raumfeld_ak4104_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 fmt, ret = 0, clk = 0; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + set_max9485_clk(MAX9485_MCLK_FREQ_122880); + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + set_max9485_clk(MAX9485_MCLK_FREQ_112896); + clk = 11289600; + break; + } + + fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF; + + /* setup the CODEC DAI */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* setup the CPU DAI */ + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, clk); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, 0, 1); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops raumfeld_ak4104_ops = { + .hw_params = raumfeld_ak4104_hw_params, +}; + +static struct snd_soc_dai_link raumfeld_spdif_dai = { + .name = "ak4104", + .stream_name = "Playback", + .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP2], + .codec_dai = &ak4104_dai, + .ops = &raumfeld_ak4104_ops, +}; + +static struct snd_soc_card snd_soc_spdif_raumfeld = { + .name = "Raumfeld S/PDIF", + .platform = &pxa2xx_soc_platform, + .dai_link = &raumfeld_spdif_dai, + .num_links = 1 +}; + +/* raumfeld_audio audio subsystem */ +static struct snd_soc_device raumfeld_line_devdata = { + .card = &snd_soc_line_raumfeld, + .codec_dev = &soc_codec_device_cs4270, +}; + +static struct snd_soc_device raumfeld_spdif_devdata = { + .card = &snd_soc_spdif_raumfeld, + .codec_dev = &soc_codec_device_ak4104, +}; + +static struct platform_device *raumfeld_audio_line_device; +static struct platform_device *raumfeld_audio_spdif_device; + +static int __init raumfeld_audio_init(void) +{ + int ret; + + if (!machine_is_raumfeld_speaker() && + !machine_is_raumfeld_connector()) + return 0; + + max9486_client = i2c_new_device(i2c_get_adapter(0), + &max9486_hwmon_info); + + if (!max9486_client) + return -ENOMEM; + + set_max9485_clk(MAX9485_MCLK_FREQ_122880); + + /* LINE */ + raumfeld_audio_line_device = platform_device_alloc("soc-audio", 0); + if (!raumfeld_audio_line_device) + return -ENOMEM; + + platform_set_drvdata(raumfeld_audio_line_device, + &raumfeld_line_devdata); + raumfeld_line_devdata.dev = &raumfeld_audio_line_device->dev; + ret = platform_device_add(raumfeld_audio_line_device); + if (ret) + platform_device_put(raumfeld_audio_line_device); + + /* no S/PDIF on Speakers */ + if (machine_is_raumfeld_speaker()) + return ret; + + /* S/PDIF */ + raumfeld_audio_spdif_device = platform_device_alloc("soc-audio", 1); + if (!raumfeld_audio_spdif_device) { + platform_device_put(raumfeld_audio_line_device); + return -ENOMEM; + } + + platform_set_drvdata(raumfeld_audio_spdif_device, + &raumfeld_spdif_devdata); + raumfeld_spdif_devdata.dev = &raumfeld_audio_spdif_device->dev; + ret = platform_device_add(raumfeld_audio_spdif_device); + if (ret) { + platform_device_put(raumfeld_audio_line_device); + platform_device_put(raumfeld_audio_spdif_device); + } + + raumfeld_enable_audio(true); + + return ret; +} + +static void __exit raumfeld_audio_exit(void) +{ + raumfeld_enable_audio(false); + + platform_device_unregister(raumfeld_audio_line_device); + + if (machine_is_raumfeld_connector()) + platform_device_unregister(raumfeld_audio_spdif_device); + + i2c_unregister_device(max9486_client); + + gpio_free(GPIO_MCLK_RESET); + gpio_free(GPIO_CODEC_RESET); + gpio_free(GPIO_SPDIF_RESET); +} + +module_init(raumfeld_audio_init); +module_exit(raumfeld_audio_exit); + +/* Module information */ +MODULE_AUTHOR("Daniel Mack <daniel(a)caiaq.de>"); +MODULE_DESCRIPTION("Raumfeld audio SoC"); +MODULE_LICENSE("GPL"); -- 1.6.5.2

2 1

[alsa-devel] Sound support for Neuros OSD2 Davinci based board.
by A. Porodko 27 Nov '09

27 Nov '09

Hello, Here is a patch for Neuros OSD2 Davinci (DM6446) based board sound support. Patch made against 2.6.32-rc6 kernel. ------------------------------------------ -- Best regards Andrey A. Porodko

3 3

[alsa-devel] [PATCH - alsa-plugins 1/2] a52 - fix 5.1 channel order with recent libavcodec
by Anssi Hannula 27 Nov '09

27 Nov '09

As of SVN r18540 libavcodec expects 5.1 channel audio with SMPTE channel order. Fix ALSA a52 plugin to use that order when built against such a libavcodec. Minor version of libavcodec was raised on the same day (Apr 17th 2009), so use that for the check. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> --- a52/pcm_a52.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/a52/pcm_a52.c b/a52/pcm_a52.c index 1abfdaa..b95d7d4 100644 --- a/a52/pcm_a52.c +++ b/a52/pcm_a52.c @@ -170,7 +170,13 @@ static int fill_data(snd_pcm_ioplug_t *io, static unsigned int ch_index[3][6] = { { 0, 1 }, { 0, 1, 2, 3 }, +#if LIBAVCODEC_VERSION_MAJOR > 52 || (LIBAVCODEC_VERSION_MAJOR == 52 && LIBAVCODEC_VERSION_MINOR >= 26) + /* current libavcodec expects SMPTE order */ + { 0, 1, 4, 5, 2, 3 }, +#else + /* libavcodec older than r18540 expects A52 order */ { 0, 4, 1, 2, 3, 5 }, +#endif }; /* flatten copy to n-channel interleaved */ dst_step = io->channels; -- 1.6.4.4

2 2

[alsa-devel] ASoC: Revert missing reset_err in wm97*.c
by Takashi Iwai 27 Nov '09

27 Nov '09

FYI, I committed the fix below to topic/asoc branch now. thanks, Takashi === From a22eaf4ce106404f6c5283da30b4d514ede964c1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai <tiwai(a)suse.de> Date: Fri, 27 Nov 2009 15:14:09 +0100 Subject: [PATCH] ASoC: Revert missing reset_err in wm97*.c The commit fe3e78e073d25308756f38019956061153267769 ASoC: Factor out snd_soc_init_card() removed the error paths that are still valid for wm97* codecs, causing the compile errors like sound/soc/codecs/wm9705.c:399: error: label 'reset_err' used but not defined sound/soc/codecs/wm9712.c:687: error: label 'reset_err' used but not defined sound/soc/codecs/wm9713.c:1237: error: label 'reset_err' used but not defined Revert the removed error path codes. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/codecs/wm9705.c | 2 ++ sound/soc/codecs/wm9712.c | 2 ++ sound/soc/codecs/wm9713.c | 2 ++ 3 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index dfffc6c..ec54c6d 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -404,6 +404,8 @@ static int wm9705_soc_probe(struct platform_device *pdev) return 0; +reset_err: + snd_soc_free_pcms(socdev); pcm_err: snd_soc_free_ac97_codec(codec); codec_err: diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 2a08722..0ac1215 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -697,6 +697,8 @@ static int wm9712_soc_probe(struct platform_device *pdev) return 0; +reset_err: + snd_soc_free_pcms(socdev); pcm_err: snd_soc_free_ac97_codec(codec); diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 00bac31..4d74ecb 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1249,6 +1249,8 @@ static int wm9713_soc_probe(struct platform_device *pdev) return 0; +reset_err: + snd_soc_free_pcms(socdev); pcm_err: snd_soc_free_ac97_codec(codec); -- 1.6.5.3

2 1

Re: [alsa-devel] Sound support for Neuros OSD2 Davinci based board.
by Mark Brown 27 Nov '09

27 Nov '09

On Fri, Nov 27, 2009 at 06:14:26PM +0500, A. Porodko wrote: > Mark Brown wrote: > > I raised the same query when Andrey sent an off-list copy to me by > > mistake - he said that in future he intends to extend the driver to > > have additional features not present on the EVM (though for me that > > could always change the || later). > Understood. Shall I change if if to if || then? Honestly I'm not that bothered, I was mostly following up to the list so that readers and the archives could see the answer to the query (I guessed you may have mistakenly replied off list).

1 0

[alsa-devel] [PATCH] ASoC: AIC23: Fixing infinite loop in resume path
by Anuj Aggarwal 27 Nov '09

27 Nov '09

This patch fixes two issues: a) Infinite loop in resume function b) Writes to non-existing registers in resume function Signed-off-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> --- sound/soc/codecs/tlv320aic23.c | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 6b24d8b..90a0264 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -625,11 +625,10 @@ static int tlv320aic23_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; - int i; u16 reg; /* Sync reg_cache with the hardware */ - for (reg = 0; reg < ARRAY_SIZE(tlv320aic23_reg); i++) { + for (reg = 0; reg < TLV320AIC23_RESET; reg++) { u16 val = tlv320aic23_read_reg_cache(codec, reg); tlv320aic23_write(codec, reg, val); } -- 1.6.2.4

4 3

[alsa-devel] [PATCH 2/3] ALSA: ARM: add Raumfeld audio support
by Daniel Mack 27 Nov '09

27 Nov '09

Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/pxa/Kconfig | 9 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/raumfeld.c | 416 ++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 427 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/raumfeld.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index dcb3181..9c3bfd9 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -117,6 +117,15 @@ config SND_SOC_ZYLONITE Say Y if you want to add support for SoC audio on the Marvell Zylonite reference platform. +config SND_SOC_RAUMFELD + tristate "SoC Audio support Raumfeld audio adapter" + depends on SND_PXA2XX_SOC && (MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR) + select SND_PXA_SOC_SSP + select SND_SOC_CS4270 + select SND_SOC_AK4104 + help + Say Y if you want to add support for SoC audio on Raumfeld devices + config SND_PXA2XX_SOC_MAGICIAN tristate "SoC Audio support for HTC Magician" depends on SND_PXA2XX_SOC && MACH_MAGICIAN diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 6e096b4..f3e08fd 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -23,6 +23,7 @@ snd-soc-zylonite-objs := zylonite.o snd-soc-magician-objs := magician.o snd-soc-mioa701-objs := mioa701_wm9713.o snd-soc-imote2-objs := imote2.o +snd-soc-raumfeld-objs := raumfeld.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o @@ -37,3 +38,4 @@ obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o +obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c new file mode 100644 index 0000000..45e3f1c --- /dev/null +++ b/sound/soc/pxa/raumfeld.c @@ -0,0 +1,416 @@ +/* + * raumfeld_audio.c -- SoC audio for Raumfeld audio devices + * + * Copyright (c) 2009 Daniel Mack <daniel(a)caiaq.de> + * + * based on code from: + * + * Wolfson Microelectronics PLC. + * Openedhand Ltd. + * Liam Girdwood <lrg(a)slimlogic.co.uk> + * Richard Purdie <richard(a)openedhand.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; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/module.h> +#include <linux/i2c.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/regulator/machine.h> +#include <linux/regulator/fixed.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/mfp-pxa3xx.h> +#include <mach/mfp-pxa300.h> +#include <asm/mach-types.h> + +#include "../codecs/cs4270.h" +#include "../codecs/ak4104.h" +#include "pxa2xx-pcm.h" +#include "pxa-ssp.h" + +#define GPIO_SPDIF_RESET MFP_PIN_GPIO38 +#define GPIO_MCLK_RESET MFP_PIN_GPIO111 +#define GPIO_CODEC_RESET MFP_PIN_GPIO120 +#define GPIO_AUDIO_VA_ENABLE MFP_PIN_GPIO124 + +static struct i2c_client *max9486_client; +static struct i2c_board_info max9486_hwmon_info = { + I2C_BOARD_INFO("max9485", 0x63), +}; + +#define MAX9485_MCLK_FREQ_112896 0x22 +#define MAX9485_MCLK_FREQ_122880 0x23 + +static void set_max9485_clk(char clk) +{ + i2c_master_send(max9486_client, &clk, 1); +} + +/* Fixed regulator for AUDIO_VA */ + +static struct regulator_consumer_supply audio_va_consumer_supply = + REGULATOR_SUPPLY("va", "soc-audio.0"); + +struct regulator_init_data audio_va_initdata = { + .consumer_supplies = &audio_va_consumer_supply, + .num_consumer_supplies = 1, + .constraints = { + .valid_ops_mask = REGULATOR_CHANGE_STATUS, + }, +}; + +static struct fixed_voltage_config audio_va_config = { + .supply_name = "audio_va", + .microvolts = 5000000, + .gpio = mfp_to_gpio(GPIO_AUDIO_VA_ENABLE), + .enable_high = 1, + .enabled_at_boot = 0, + .init_data = &audio_va_initdata, +}; + +static struct platform_device audio_va_device = { + .name = "reg-fixed-voltage", + .id = 0, + .dev = { + .platform_data = &audio_va_config, + }, +}; + +/* Dummy supplies for Codec's VD/VLC */ + +static struct regulator_consumer_supply audio_dummy_supplies[] = { + REGULATOR_SUPPLY("vd", "soc-audio.0"), + REGULATOR_SUPPLY("vlc", "soc-audio.0"), +}; + +struct regulator_init_data audio_dummy_initdata = { + .consumer_supplies = audio_dummy_supplies, + .num_consumer_supplies = ARRAY_SIZE(audio_dummy_supplies), + .constraints = { + .valid_ops_mask = REGULATOR_CHANGE_STATUS, + }, +}; + +static struct fixed_voltage_config audio_dummy_config = { + .supply_name = "audio_vd", + .microvolts = 3300000, + .gpio = -1, + .init_data = &audio_dummy_initdata, +}; + +static struct platform_device audio_supply_dummy_device = { + .name = "reg-fixed-voltage", + .id = 1, + .dev = { + .platform_data = &audio_dummy_config, + }, +}; + +static struct platform_device *regulator_devices[] = { + &audio_va_device, + &audio_supply_dummy_device, +}; + +static void raumfeld_enable_audio(bool en) +{ + if (en) { + gpio_set_value(mfp_to_gpio(GPIO_MCLK_RESET), 1); + + /* wait some time to let the clocks become stable */ + msleep(100); + + gpio_set_value(mfp_to_gpio(GPIO_SPDIF_RESET), 1); + gpio_set_value(mfp_to_gpio(GPIO_CODEC_RESET), 1); + } else { + gpio_set_value(mfp_to_gpio(GPIO_MCLK_RESET), 0); + gpio_set_value(mfp_to_gpio(GPIO_SPDIF_RESET), 0); + gpio_set_value(mfp_to_gpio(GPIO_CODEC_RESET), 0); + } +} + +/* CS4270 */ +static int raumfeld_cs4270_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + set_max9485_clk(MAX9485_MCLK_FREQ_112896); + + return snd_soc_dai_set_sysclk(codec_dai, 0, 11289600, 0); +} + +static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int fmt, clk = 0; + int ret = 0; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + set_max9485_clk(MAX9485_MCLK_FREQ_122880); + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + set_max9485_clk(MAX9485_MCLK_FREQ_112896); + clk = 11289600; + break; + } + + fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; + + /* setup the CODEC DAI */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, 0); + if (ret < 0) + return ret; + + /* setup the CPU DAI */ + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, clk); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, 0, 1); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops raumfeld_cs4270_ops = { + .startup = raumfeld_cs4270_startup, + .hw_params = raumfeld_cs4270_hw_params, +}; + +static int raumfeld_line_suspend(struct platform_device *pdev, pm_message_t state) +{ + raumfeld_enable_audio(false); + return 0; +} + +static int raumfeld_line_resume(struct platform_device *pdev) +{ + raumfeld_enable_audio(true); + return 0; +} + +static struct snd_soc_dai_link raumfeld_line_dai = { + .name = "CS4270", + .stream_name = "CS4270", + .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP1], + .codec_dai = &cs4270_dai, + .ops = &raumfeld_cs4270_ops, +}; + +static struct snd_soc_card snd_soc_line_raumfeld = { + .name = "Raumfeld analog", + .platform = &pxa2xx_soc_platform, + .dai_link = &raumfeld_line_dai, + .suspend_post = raumfeld_line_suspend, + .resume_pre = raumfeld_line_resume, + .num_links = 1, +}; + + +/* AK4104 */ + +static int raumfeld_ak4104_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 fmt, ret = 0, clk = 0; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + set_max9485_clk(MAX9485_MCLK_FREQ_122880); + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + set_max9485_clk(MAX9485_MCLK_FREQ_112896); + clk = 11289600; + break; + } + + fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF; + + /* setup the CODEC DAI */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* setup the CPU DAI */ + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, clk); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, 0, 1); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops raumfeld_ak4104_ops = { + .hw_params = raumfeld_ak4104_hw_params, +}; + +static int raumfeld_ak4104_init(struct snd_soc_codec *codec) +{ + return 0; +} + +static struct snd_soc_dai_link raumfeld_spdif_dai = { + .name = "ak4104", + .stream_name = "Playback", + .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP2], + .codec_dai = &ak4104_dai, + .init = raumfeld_ak4104_init, + .ops = &raumfeld_ak4104_ops, +}; + +static struct snd_soc_card snd_soc_spdif_raumfeld = { + .name = "Raumfeld S/PDIF", + .platform = &pxa2xx_soc_platform, + .dai_link = &raumfeld_spdif_dai, + .num_links = 1 +}; + +/* raumfeld_audio audio subsystem */ +static struct snd_soc_device raumfeld_line_devdata = { + .card = &snd_soc_line_raumfeld, + .codec_dev = &soc_codec_device_cs4270, +}; + +static struct snd_soc_device raumfeld_spdif_devdata = { + .card = &snd_soc_spdif_raumfeld, + .codec_dev = &soc_codec_device_ak4104, +}; + +static struct platform_device *raumfeld_audio_line_device; +static struct platform_device *raumfeld_audio_spdif_device; + +static int __init raumfeld_audio_init(void) +{ + int ret; + + if (!machine_is_raumfeld_speaker() && + !machine_is_raumfeld_connector()) + return 0; + + max9486_client = i2c_new_device(i2c_get_adapter(0), + &max9486_hwmon_info); + + if (!max9486_client) + return -ENOMEM; + + set_max9485_clk(MAX9485_MCLK_FREQ_122880); + + platform_add_devices(regulator_devices, ARRAY_SIZE(regulator_devices)); + + /* LINE */ + raumfeld_audio_line_device = platform_device_alloc("soc-audio", 0); + if (!raumfeld_audio_line_device) + return -ENOMEM; + + platform_set_drvdata(raumfeld_audio_line_device, + &raumfeld_line_devdata); + raumfeld_line_devdata.dev = &raumfeld_audio_line_device->dev; + ret = platform_device_add(raumfeld_audio_line_device); + if (ret) + platform_device_put(raumfeld_audio_line_device); + + /* no S/PDIF on Speakers */ + if (machine_is_raumfeld_speaker()) + return ret; + + /* S/PDIF */ + raumfeld_audio_spdif_device = platform_device_alloc("soc-audio", 1); + if (!raumfeld_audio_spdif_device) { + platform_device_put(raumfeld_audio_line_device); + return -ENOMEM; + } + + platform_set_drvdata(raumfeld_audio_spdif_device, + &raumfeld_spdif_devdata); + raumfeld_spdif_devdata.dev = &raumfeld_audio_spdif_device->dev; + ret = platform_device_add(raumfeld_audio_spdif_device); + if (ret) { + platform_device_put(raumfeld_audio_line_device); + platform_device_put(raumfeld_audio_spdif_device); + } + + raumfeld_enable_audio(true); + + return ret; +} + +static void __exit raumfeld_audio_exit(void) +{ + raumfeld_enable_audio(false); + + platform_device_unregister(raumfeld_audio_line_device); + + if (machine_is_raumfeld_connector()) + platform_device_unregister(raumfeld_audio_spdif_device); + + i2c_unregister_device(max9486_client); + + platform_device_put(&audio_va_device); + platform_device_put(&audio_supply_dummy_device); + + gpio_free(mfp_to_gpio(GPIO_MCLK_RESET)); + gpio_free(mfp_to_gpio(GPIO_CODEC_RESET)); + gpio_free(mfp_to_gpio(GPIO_SPDIF_RESET)); +} + +module_init(raumfeld_audio_init); +module_exit(raumfeld_audio_exit); + +/* Module information */ +MODULE_AUTHOR("Daniel Mack <daniel(a)caiaq.de>"); +MODULE_DESCRIPTION("Raumfeld audio SoC"); +MODULE_LICENSE("GPL"); -- 1.6.5.2

2 2

[alsa-devel] ugly sound output (intel-hda) [was: mmotm 2009-11-24-16-47 uploaded]
by Jiri Slaby 27 Nov '09

27 Nov '09

On 11/25/2009 01:47 AM, akpm(a)linux-foundation.org wrote: > The mm-of-the-moment snapshot 2009-11-24-16-47 has been uploaded to Hi, when I boot into that kernel, I hear, since the boot phase, short scratches repeatedly every (just a guess) 100 ms. # cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xffa74000 irq 29 # cat /proc/asound/card0/codec#2 Codec: SigmaTel STAC9274D Address: 2 Function Id: 0x1 Vendor Id: 0x83847621 Subsystem Id: 0x00000100 Revision Id: 0x100201 No Modem Function Group found Default PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: ofs=0x00, nsteps=0x0e, stepsize=0x05, mute=0 Default Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x02, mute=1 GPIO: io=3, o=0, i=0, unsolicited=1, wake=1 IO[0]: enable=1, dir=1, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Analog Loopback: 0x00 Node 0x02 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Control: name="Front Playback Volume", index=0, device=0 Control: name="Front Playback Switch", index=0, device=0 Device: name="STAC92xx Analog", type="Audio", device=0 Amp-Out caps: N/A Amp-Out vals: [0x67 0x67] Converter: stream=5, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x03 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Control: name="Side Playback Volume", index=0, device=0 Control: name="Side Playback Switch", index=0, device=0 Amp-Out caps: N/A Amp-Out vals: [0x67 0x67] Converter: stream=5, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x04 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Control: name="Center Playback Volume", index=0, device=0 Control: name="Center Playback Switch", index=0, device=0 Control: name="LFE Playback Volume", index=0, device=0 Control: name="LFE Playback Switch", index=0, device=0 Control: name="Swap Center/LFE Playback Switch", index=0, device=0 Amp-Out caps: N/A Amp-Out vals: [0x67 0x67] Converter: stream=5, channel=4 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x05 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Control: name="Surround Playback Volume", index=0, device=0 Control: name="Surround Playback Switch", index=0, device=0 Amp-Out caps: N/A Amp-Out vals: [0x67 0x67] Converter: stream=5, channel=2 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x06 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Control: name="Headphone Playback Volume", index=0, device=0 Control: name="Headphone Playback Switch", index=0, device=0 Amp-Out caps: N/A Amp-Out vals: [0x67 0x67] Converter: stream=5, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x07 [Audio Input] wcaps 0x1d0541: Stereo Device: name="STAC92xx Analog", type="Audio", device=0 Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x1b Processing caps: benign=0, ncoeff=0 Node 0x08 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x1c Processing caps: benign=0, ncoeff=0 Node 0x09 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x1d Processing caps: benign=0, ncoeff=0 Node 0x0a [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0000173f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x02214020: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=01, enabled=1 Connection: 3 0x02 0x03 0x06* Node 0x0b [Pin Complex] wcaps 0x400181: Stereo Control: name="Front Mic Jack Mode", index=0, device=0 Pincap 0x0000173f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x02a19080: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x8, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=03, enabled=1 Connection: 3 0x02* 0x03 0x06 Node 0x0c [Pin Complex] wcaps 0x400181: Stereo Control: name="Line Jack Mode", index=0, device=0 Pincap 0x00001737: IN OUT Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x0181304e: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x4, Sequence = 0xe Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=04, enabled=1 Connection: 1 0x03 Node 0x0d [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0000173f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x02 Node 0x0e [Pin Complex] wcaps 0x400181: Stereo Control: name="Mic Jack Mode", index=0, device=0 Pincap 0x00001737: IN OUT Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x01a19040: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Connection: 1 0x04 Node 0x0f [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00001737: IN OUT Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x01011012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x40: OUT VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x05 Node 0x10 [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00000037: IN OUT Detect Trigger ImpSense Pin Default 0x01016011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x04 Node 0x11 [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00000037: IN OUT Detect Trigger ImpSense Pin Default 0x01012014: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x03 Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x503301f0: [N/A] CD at Int N/A Conn = ATAPI, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x13 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x50a001f0: [N/A] Mic at Int N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x14 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x50a001f0: [N/A] Mic at Int N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x15 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Control: name="Mux Capture Volume", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 9 0x0e* 0x12 0x0f 0x0b 0x0c 0x0d 0x0a 0x10 0x11 Node 0x16 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Control: name="Mux Capture Volume", index=1, device=0 Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 9 0x0e* 0x12 0x0f 0x0b 0x0c 0x0d 0x0a 0x10 0x11 Node 0x17 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Control: name="Mux Capture Volume", index=2, device=0 Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 9 0x0e* 0x12 0x0f 0x0b 0x0c 0x0d 0x0a 0x10 0x11 Node 0x18 [Audio Selector] wcaps 0x300103: Stereo Amp-In Control: name="Capture Volume", index=0, device=0 Amp-In caps: N/A Amp-In vals: [0x08 0x08] Connection: 1 0x15 Node 0x19 [Audio Selector] wcaps 0x300103: Stereo Amp-In Control: name="Capture Volume", index=1, device=0 Amp-In caps: N/A Amp-In vals: [0x00 0x00] Connection: 1 0x16 Node 0x1a [Audio Selector] wcaps 0x300103: Stereo Amp-In Control: name="Capture Volume", index=2, device=0 Amp-In caps: N/A Amp-In vals: [0x00 0x00] Connection: 1 0x17 Node 0x1b [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L Control: name="Capture Switch", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Connection: 3 0x18* 0x13 0x14 Node 0x1c [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L Control: name="Capture Switch", index=1, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Connection: 3 0x19* 0x13 0x14 Node 0x1d [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L Control: name="Capture Switch", index=2, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Connection: 3 0x1a* 0x13 0x14 Node 0x1e [Audio Output] wcaps 0x40211: 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="STAC92xx Digital", type="SPDIF", device=1 Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Node 0x1f [Vendor Defined Widget] wcaps 0xf30201: Stereo Digital Delay: 3 samples Node 0x20 [Audio Input] wcaps 0x140311: Stereo Digital Device: name="STAC92xx Digital", type="SPDIF", device=1 Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Connection: 1 0x22 Node 0x21 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x01442170: [Jack] SPDIF Out at Ext Rear Conn = RCA, Color = Grey DefAssociation = 0x7, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 5 0x1e* 0x1f 0x1b 0x1c 0x1d Node 0x22 [Pin Complex] wcaps 0x430681: Stereo Digital Pincap 0x00010024: IN EAPD Detect EAPD 0x0: Pin Default 0x81c42090: [Fixed] SPDIF In at Ext Rear Conn = RCA, Color = Grey DefAssociation = 0x9, Sequence = 0x0 Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Delay: 3 samples Node 0x23 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=0 Amp-Out vals: [0x00] Node 0x24 [Volume Knob Widget] wcaps 0x600000: Mono Volume-Knob: delta=1, steps=127, direct=1, val=127 Connection: 5 0x02 0x03 0x04 0x05 0x06 It's a regression against, I think, 2009-11-13-19-59. thanks, -- js

2 5

[alsa-devel] [PATCH] ASoC: AM3517: Fix AIC23 suspend/resume hang
by Anuj Aggarwal 27 Nov '09

27 Nov '09

System hang is observed While trying to do suspend. It was because of an infinite loop in the AIC23 resume path which was trying to restore AIC23 register values from the register cache. This patch fixes the problem by correcting the resume path and properly activating/deactivating the digital interface while doing the suspend / off transitions. Signed-off-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> --- sound/soc/codecs/tlv320aic23.c | 16 +++------------- 1 files changed, 3 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 6b24d8b..cabe60c 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -565,13 +565,12 @@ static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - /* everything off except vref/vmid, */ - tlv320aic23_write(codec, TLV320AIC23_PWR, reg | 0x0040); + /* Activate the digital interface */ + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1); break; case SND_SOC_BIAS_OFF: - /* everything off, dac mute, inactive */ + /* Deactivate the digital interface */ tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); - tlv320aic23_write(codec, TLV320AIC23_PWR, 0xffff); break; } codec->bias_level = level; @@ -615,7 +614,6 @@ static int tlv320aic23_suspend(struct platform_device *pdev, struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; - tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; @@ -625,14 +623,6 @@ static int tlv320aic23_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; - int i; - u16 reg; - - /* Sync reg_cache with the hardware */ - for (reg = 0; reg < ARRAY_SIZE(tlv320aic23_reg); i++) { - u16 val = tlv320aic23_read_reg_cache(codec, reg); - tlv320aic23_write(codec, reg, val); - } tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); tlv320aic23_set_bias_level(codec, codec->suspend_bias_level); -- 1.6.2.4

5 16

[alsa-devel] Problem with ASUSTeK Computer Inc. Device 8398
by Bráulio B O Bhavamitra 27 Nov '09

27 Nov '09

Hello I'm using Ubuntu Karmic on a Asus T91 netbook. The sounds works except for the internal MIC so I have installed linux-backports-modules-alsa-karmic-generic (linux-backports-modules-alsa-2.6.31-15-generic). Now the problem is that unpredictably the sound stop working, that is, the apps play but no sound is really played. Looking at dmesg reveals lots of errors (see below). lspci output is also attached. Should I provice more information? Any devel version I can use/test? Regards, bráulio $ dmesg [ 240.404023] ALSA hda_intel.c:699: azx_get_response timeout, switching to polling mode: last cmd=0x015f0900 [ 241.408023] ALSA hda_intel.c:723: hda_intel: azx_get_response timeout, switching to single_cmd mode: last cmd=0x015f0900 [ 241.408097] ALSA hda_intel.c:647: spurious response 0x0:0x0, last cmd=0x15f0900 [ 241.408116] ALSA hda_intel.c:647: spurious response 0x0:0x0, last cmd=0x1470740 [ 241.408198] ALSA hda_intel.c:647: spurious response 0x0:0x0, last cmd=0x1470740 [ 241.408281] ALSA hda_intel.c:647: spurious response 0x0:0x0, last cmd=0x18f0900 [ 241.408303] ALSA hda_intel.c:647: spurious response 0x0:0x0, last cmd=0x2370105 [ 241.408386] ALSA hda_intel.c:647: spurious response 0x0:0x0, last cmd=0x23a02b [ 241.408469] ALSA hda_intel.c:647: spurious response 0x0:0x0, last cmd=0x23902b [ 241.408490] ALSA hda_intel.c:647: spurious response 0x0:0x0, last cmd=0x1539000 [ 241.408573] ALSA hda_intel.c:647: spurious response 0x0:0x0, last cmd=0x1539000 [ 241.408656] ALSA hda_intel.c:647: spurious response 0x0:0x0, last cmd=0x33a01f [ 241.408677] ALSA hda_intel.c:647: spurious response 0x0:0x0, last cmd=0x33901f [ 241.408760] ALSA hda_intel.c:647: spurious response 0x0:0x0, last cmd=0x835014 (continues) $ lspci -v 00:1b.0 Audio device: Intel Corporation System Controller Hub (SCH Poulsbo) HD Audio Controller (rev 07) Subsystem: ASUSTeK Computer Inc. Device 8398 Flags: bus master, fast devsel, latency 0, IRQ 23 Memory at f3f38000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel

1 0

[alsa-devel] [PATCH 1/2 v2] ASoC: sh: fsi: Add runtime PM support
by Kuninori Morimoto 27 Nov '09

27 Nov '09

This patch add support runtime PM. Driver callbacks for Runtime PM are empty because the device registers are always re-initialized after pm_runtime_get_sync(). The Runtime PM functions replaces the clock framework module stop bit handling in this driver. Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- v1 -> v2 o add pm_runtime_disable if failed sound/soc/sh/fsi.c | 40 ++++++++++++++++++++++++++-------------- 1 files changed, 26 insertions(+), 14 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index e1a3d1a..e2f70ac 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -17,7 +17,7 @@ #include <linux/platform_device.h> #include <linux/delay.h> #include <linux/list.h> -#include <linux/clk.h> +#include <linux/pm_runtime.h> #include <linux/io.h> #include <sound/core.h> #include <sound/pcm.h> @@ -105,7 +105,6 @@ struct fsi_priv { struct fsi_master { void __iomem *base; int irq; - struct clk *clk; struct fsi_priv fsia; struct fsi_priv fsib; struct sh_fsi_platform_info *info; @@ -559,7 +558,7 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, int is_master; int ret = 0; - clk_enable(master->clk); + pm_runtime_get_sync(dai->dev); /* CKG1 */ data = is_play ? (1 << 0) : (1 << 4); @@ -674,7 +673,7 @@ static void fsi_dai_shutdown(struct snd_pcm_substream *substream, fsi_irq_disable(fsi, is_play); fsi_clk_ctrl(fsi, 0); - clk_disable(master->clk); + pm_runtime_put_sync(dai->dev); } static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, @@ -872,7 +871,6 @@ EXPORT_SYMBOL_GPL(fsi_soc_platform); static int fsi_probe(struct platform_device *pdev) { struct resource *res; - char clk_name[8]; unsigned int irq; int ret; @@ -903,14 +901,9 @@ static int fsi_probe(struct platform_device *pdev) master->fsia.base = master->base; master->fsib.base = master->base + 0x40; - /* FSI is based on SPU mstp */ - snprintf(clk_name, sizeof(clk_name), "spu%d", pdev->id); - master->clk = clk_get(NULL, clk_name); - if (IS_ERR(master->clk)) { - dev_err(&pdev->dev, "cannot get %s mstp\n", clk_name); - ret = -EIO; - goto exit_iounmap; - } + pm_suspend_ignore_children(&pdev->dev, true); + pm_runtime_enable(&pdev->dev); + pm_runtime_resume(&pdev->dev); fsi_soc_dai[0].dev = &pdev->dev; fsi_soc_dai[1].dev = &pdev->dev; @@ -935,6 +928,7 @@ exit_free_irq: free_irq(irq, master); exit_iounmap: iounmap(master->base); + pm_runtime_disable(&pdev->dev); exit_kfree: kfree(master); master = NULL; @@ -947,7 +941,7 @@ static int fsi_remove(struct platform_device *pdev) snd_soc_unregister_dais(fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai)); snd_soc_unregister_platform(&fsi_soc_platform); - clk_put(master->clk); + pm_runtime_disable(&pdev->dev); free_irq(master->irq, master); @@ -957,9 +951,27 @@ static int fsi_remove(struct platform_device *pdev) return 0; } +static int fsi_runtime_nop(struct device *dev) +{ + /* Runtime PM callback shared between ->runtime_suspend() + * and ->runtime_resume(). Simply returns success. + * + * This driver re-initializes all registers after + * pm_runtime_get_sync() anyway so there is no need + * to save and restore registers here. + */ + return 0; +} + +static struct dev_pm_ops fsi_pm_ops = { + .runtime_suspend = fsi_runtime_nop, + .runtime_resume = fsi_runtime_nop, +}; + static struct platform_driver fsi_driver = { .driver = { .name = "sh_fsi", + .pm = &fsi_pm_ops, }, .probe = fsi_probe, .remove = fsi_remove, -- 1.6.3.3

5 15

[alsa-devel] [PATCH] opti-miro: fix OOPS if hardware is not detected
by Krzysztof Helt 27 Nov '09

27 Nov '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> If a hardware is not detected there is a kernel crash due to not initialized snd_miro->aci pointer. This pointer is initialized after detection of the opti (miro) chip. This bug was introduced by patches to expose ACI mikser outside the snd-miro driver. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/opti9xx/miro.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index 40b64cd..e374869 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -1229,7 +1229,8 @@ static void snd_card_miro_free(struct snd_card *card) struct snd_miro *miro = card->private_data; release_and_free_resource(miro->res_aci_port); - miro->aci->aci_port = 0; + if (miro->aci) + miro->aci->aci_port = 0; release_and_free_resource(miro->res_mc_base); } -- 1.6.4 ---------------------------------------------------------------------- Wygraj pobyt w Alpach dla ca�ej rodziny Kliknij >>> http://link.interia.pl/f2446

2 1

[alsa-devel] [PATCH 0/5] PCM mmap (temporary) fixes for non-coherent architectures
by Takashi Iwai 27 Nov '09

27 Nov '09

Hi, this is a patchset to fix Oopses from ALSA PCM core with mmap on MIPS and PPC non-coherent architectures. This contains also a clean-up for ARM mmap. In this patch series, I don't try to port dma_mmap_coherent() to these architectures yet, but just put ugly ifdefs in the ALSA core side. Understand that this is a first step forward to a more cleaner solution, and the purpose right now is just to fix up long-standing Oops. I'll try to push dma_mmap_coherent() patches for the next kernel once again after this gets merged. Since the changes are minimal, I'd like to put them in ASAP for 2.6.33. Please review and give feedbacks if any problem is found. Thanks, Takashi

4 13

[alsa-devel] audio pm on omap - resume tik sound
by Ryan 27 Nov '09

27 Nov '09

Hello, When audio resumes from suspend i hear a tik sound on my speakers. Why is this sound produced and is there a fix somewhere for this. Regards, ryan

1 2

[alsa-devel] [PATCH] ASoC: tlv320dac33: Change RT wq to singlethread wq
by Peter Ujfalusi 27 Nov '09

27 Nov '09

RT workqueue is going away in the near future, replace it with singlethread wq for now, which is still supported. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- Hello, I believe this patch fixes the linux-next breakage (in the tlv320dac33 codec driver) caused by the removal of the RT workqueue in the wq tree. sound/soc/codecs/tlv320dac33.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 2a013e4..9c8903d 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1118,7 +1118,8 @@ static int dac33_i2c_probe(struct i2c_client *client, } if (dac33->irq != -1) { /* Setup work queue */ - dac33->dac33_wq = create_rt_workqueue("tlv320dac33"); + dac33->dac33_wq = + create_singlethread_workqueue("tlv320dac33"); if (dac33->dac33_wq == NULL) { free_irq(dac33->irq, &dac33->codec); ret = -ENOMEM; -- 1.6.5.3

3 3

[alsa-devel] tea575x-tuner: fix mute
by Ondrej Zary 26 Nov '09

26 Nov '09

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

1 0

[alsa-devel] [PATCH 16/17] ALSA: ARM: add Raumfeld audio support
by Daniel Mack 26 Nov '09

26 Nov '09

Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/pxa/Kconfig | 9 ++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/raumfeld.c | 338 ++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 349 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/raumfeld.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index dcb3181..9c3bfd9 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -117,6 +117,15 @@ config SND_SOC_ZYLONITE Say Y if you want to add support for SoC audio on the Marvell Zylonite reference platform. +config SND_SOC_RAUMFELD + tristate "SoC Audio support Raumfeld audio adapter" + depends on SND_PXA2XX_SOC && (MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR) + select SND_PXA_SOC_SSP + select SND_SOC_CS4270 + select SND_SOC_AK4104 + help + Say Y if you want to add support for SoC audio on Raumfeld devices + config SND_PXA2XX_SOC_MAGICIAN tristate "SoC Audio support for HTC Magician" depends on SND_PXA2XX_SOC && MACH_MAGICIAN diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 6e096b4..f3e08fd 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -23,6 +23,7 @@ snd-soc-zylonite-objs := zylonite.o snd-soc-magician-objs := magician.o snd-soc-mioa701-objs := mioa701_wm9713.o snd-soc-imote2-objs := imote2.o +snd-soc-raumfeld-objs := raumfeld.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o @@ -37,3 +38,4 @@ obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o +obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c new file mode 100644 index 0000000..7efc30a --- /dev/null +++ b/sound/soc/pxa/raumfeld.c @@ -0,0 +1,338 @@ +/* + * raumfeld_audio.c -- SoC audio for Raumfeld audio devices + * + * Copyright (c) 2009 Daniel Mack <daniel(a)caiaq.de> + * + * based on code from: + * + * Wolfson Microelectronics PLC. + * Openedhand Ltd. + * Liam Girdwood <lrg(a)slimlogic.co.uk> + * Richard Purdie <richard(a)openedhand.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; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/module.h> +#include <linux/i2c.h> +#include <linux/delay.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <asm/mach-types.h> +#include <mach/mfp-pxa3xx.h> +#include <mach/mfp-pxa300.h> + +#include "../codecs/cs4270.h" +#include "../codecs/ak4104.h" +#include "pxa2xx-pcm.h" +#include "pxa-ssp.h" + +extern void raumfeld_enable_audio(bool en); + +static struct i2c_client *max9486_client; +static struct i2c_board_info max9486_hwmon_info = { + I2C_BOARD_INFO("max9485", 0x63), +}; + +/* set_max9485_clk() - call with + * clk = 0 for 11.2896 MHz + * clk = 1 for 12.2880 MHz + */ + +static void set_max9485_clk(int clk) +{ + char buf = clk ? 0x23 : 0x22; + i2c_master_send(max9486_client, &buf, 1); +} + +/* CS4270 */ +static int raumfeld_cs4270_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + set_max9485_clk(0); + + return snd_soc_dai_set_sysclk(codec_dai, 0, 11289600, 0); +} + +static void raumfeld_cs4270_shutdown(struct snd_pcm_substream *substream) +{ +} + +static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int fmt, clk = 0; + int ret = 0; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + set_max9485_clk(1); + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + set_max9485_clk(0); + clk = 11289600; + break; + } + + fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; + + /* setup the CODEC DAI */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, 0); + if (ret < 0) + return ret; + + /* setup the CPU DAI */ + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, clk); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, 0, 1); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops raumfeld_cs4270_ops = { + .startup = raumfeld_cs4270_startup, + .hw_params = raumfeld_cs4270_hw_params, + .shutdown = raumfeld_cs4270_shutdown, +}; + +static int raumfeld_cs4270_init(struct snd_soc_codec *codec) +{ + return 0; +} + +static int raumfeld_line_suspend(struct platform_device *pdev, pm_message_t state) +{ + raumfeld_enable_audio(false); + return 0; +} + +static int raumfeld_line_resume(struct platform_device *pdev) +{ + raumfeld_enable_audio(true); + return 0; +} + +static struct snd_soc_dai_link raumfeld_line_dai = { + .name = "CS4270", + .stream_name = "CS4270", + .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP1], + .codec_dai = &cs4270_dai, + .init = raumfeld_cs4270_init, + .ops = &raumfeld_cs4270_ops, +}; + +static struct snd_soc_card snd_soc_line_raumfeld = { + .name = "Raumfeld analog", + .platform = &pxa2xx_soc_platform, + .dai_link = &raumfeld_line_dai, + .suspend_post = raumfeld_line_suspend, + .resume_pre = raumfeld_line_resume, + .num_links = 1, +}; + + +/* AK4104 */ + +static int raumfeld_ak4104_startup(struct snd_pcm_substream *substream) +{ + return 0; +} + +static void raumfeld_ak4104_shutdown(struct snd_pcm_substream *substream) +{ +} + +static int raumfeld_ak4104_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 fmt, ret = 0, clk = 0; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + set_max9485_clk(1); + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + set_max9485_clk(0); + clk = 11289600; + break; + } + + fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF; + + /* setup the CODEC DAI */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* setup the CPU DAI */ + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, clk); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, 0, 1); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops raumfeld_ak4104_ops = { + .startup = raumfeld_ak4104_startup, + .hw_params = raumfeld_ak4104_hw_params, + .shutdown = raumfeld_ak4104_shutdown, +}; + +static int raumfeld_ak4104_init(struct snd_soc_codec *codec) +{ + return 0; +} + +static struct snd_soc_dai_link raumfeld_spdif_dai = { + .name = "ak4104", + .stream_name = "Playback", + .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP2], + .codec_dai = &ak4104_dai, + .init = raumfeld_ak4104_init, + .ops = &raumfeld_ak4104_ops, +}; + +static struct snd_soc_card snd_soc_spdif_raumfeld = { + .name = "Raumfeld S/PDIF", + .platform = &pxa2xx_soc_platform, + .dai_link = &raumfeld_spdif_dai, + .num_links = 1 +}; + +/* raumfeld_audio audio subsystem */ +static struct snd_soc_device raumfeld_line_devdata = { + .card = &snd_soc_line_raumfeld, + .codec_dev = &soc_codec_device_cs4270, +}; + +static struct snd_soc_device raumfeld_spdif_devdata = { + .card = &snd_soc_spdif_raumfeld, + .codec_dev = &soc_codec_device_ak4104, +}; + +static struct platform_device *raumfeld_audio_line_device; +static struct platform_device *raumfeld_audio_spdif_device; + +static int __init raumfeld_audio_init(void) +{ + int ret; + + if (!machine_is_raumfeld_speaker() && + !machine_is_raumfeld_connector()) + return 0; + + max9486_client = i2c_new_device(i2c_get_adapter(0), + &max9486_hwmon_info); + + if (!max9486_client) + return -ENOMEM; + + set_max9485_clk(1); + + /* LINE */ + raumfeld_audio_line_device = platform_device_alloc("soc-audio", 0); + if (!raumfeld_audio_line_device) + return -ENOMEM; + + platform_set_drvdata(raumfeld_audio_line_device, + &raumfeld_line_devdata); + raumfeld_line_devdata.dev = &raumfeld_audio_line_device->dev; + ret = platform_device_add(raumfeld_audio_line_device); + if (ret) + platform_device_put(raumfeld_audio_line_device); + + if (machine_is_raumfeld_speaker()) + return ret; + + /* S/PDIF */ + raumfeld_audio_spdif_device = platform_device_alloc("soc-audio", 1); + if (!raumfeld_audio_spdif_device) { + platform_device_put(raumfeld_audio_line_device); + return -ENOMEM; + } + + platform_set_drvdata(raumfeld_audio_spdif_device, + &raumfeld_spdif_devdata); + raumfeld_spdif_devdata.dev = &raumfeld_audio_spdif_device->dev; + ret = platform_device_add(raumfeld_audio_spdif_device); + if (ret) { + platform_device_put(raumfeld_audio_line_device); + platform_device_put(raumfeld_audio_spdif_device); + } + + return ret; +} + +static void __exit raumfeld_audio_exit(void) +{ + platform_device_unregister(raumfeld_audio_line_device); + + if (machine_is_raumfeld_connector()) + platform_device_unregister(raumfeld_audio_spdif_device); + + i2c_unregister_device(max9486_client); +} + +module_init(raumfeld_audio_init); +module_exit(raumfeld_audio_exit); + +/* Module information */ +MODULE_AUTHOR("Daniel Mack <daniel(a)caiaq.de>"); +MODULE_DESCRIPTION("Raumfeld audio SoC"); +MODULE_LICENSE("GPL"); -- 1.6.5.2

2 4

[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 26 Nov '09

26 Nov '09

The following changes since commit 97cef58521288ade8d957e4d568a3d409dea7d64: Mark Brown (1): Merge branch 'for-2.6.32' into for-2.6.33 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Daniel J Blueman (1): mfd: twl4030: fix ELF section mismatch... Mark Brown (2): Merge branch 'twl4030-mfd' into for-2.6.33 ASoC: Add BCLK calculation utility for TDM mode too Peter Ujfalusi (1): ASoC: tlv320dac33: Change RT wq to singlethread wq drivers/mfd/twl4030-core.c | 2 +- include/sound/soc.h | 1 + sound/soc/codecs/tlv320dac33.c | 3 ++- sound/soc/soc-utils.c | 6 ++++++ 4 files changed, 10 insertions(+), 2 deletions(-)

2 1

[alsa-devel] Conexant CX20549 (Venice) PCM volume to loud
by Cristian KLEIN 26 Nov '09

26 Nov '09

Hello, Could anybody please have a look at this bug: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4792 Thanks in advance, Cristi.

3 3

[alsa-devel] [PATCH] MIPS: Fixups of ALSA memory maps
by Wu Zhangjin 26 Nov '09

26 Nov '09

Hi, All Seems this is MIPS specific, but it's not that easy to move this patch into the arch/mips part, So, any better solution? Thanks & Regards, Wu Zhangjin ------------------------ The user application mmap audio dma regions must be dma-coherent. This patch fix it. Without this patch, artsd will fail on boot, and mplayer will exit with "Segmentation fault". (this happens on YeeLoong netbook, fuloong2f mini pc with snd_cs5535 audio card) This is originally from the to-mips branch of http://dev.lemote.com/code/linux_loongson, and contributed by Yanhua from Lemote Inc. Reported-by: qiaochong <qiaochong(a)gmail.com> Signed-off-by: Wu Zhangjin <wuzj(a)lemote.com> --- sound/core/pcm_native.c | 9 +++++++++ 1 files changed, 9 insertions(+), 0 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index ab73edf..2779b9a 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3087,7 +3087,11 @@ static int snd_pcm_mmap_data_fault(struct vm_area_struct *area, return VM_FAULT_SIGBUS; } else { vaddr = runtime->dma_area + offset; +#if defined(__mips__) && defined(CONFIG_DMA_NONCOHERENT) + page = virt_to_page(CAC_ADDR(vaddr)); +#else page = virt_to_page(vaddr); +#endif } get_page(page); vmf->page = page; @@ -3202,6 +3206,11 @@ static int snd_pcm_mmap(struct file *file, struct vm_area_struct *area) if (PCM_RUNTIME_CHECK(substream)) return -ENXIO; +#if defined(__mips__) && defined(CONFIG_DMA_NONCOHERENT) + /* all mmap using uncached mode */ + area->vm_page_prot = pgprot_noncached(area->vm_page_prot); + area->vm_flags |= (VM_RESERVED | VM_IO); +#endif offset = area->vm_pgoff << PAGE_SHIFT; switch (offset) { case SNDRV_PCM_MMAP_OFFSET_STATUS: -- 1.6.2.1

3 12

Re: [alsa-devel] Patch for Neuros OSD2 board sound support
by A. Porodko 26 Nov '09

26 Nov '09

Sure. Here it is. Liam Girdwood wrote: > Hi Andrey, > > Could you resend and also CC the alsa-devel mailing list. > > Thanks > > Liam > > On Thu, 2009-11-26 at 16:30 +0500, A. Porodko wrote: > >> Hello, >> >> I would like to propose a patch for Neuros OSD2 (davinci SOC DM6446) >> based board. >> Would you be so kind to consider it, please? Corresponded patch for >> board code I've already submitted to davinci group. >> Thank you in advance. >> >> > > > -- Best regards Andrey A. Porodko

1 0

[alsa-devel] AK4671 pin sharing
by Misael Lopez 26 Nov '09

26 Nov '09

I'm interested in adding some functionalities to ak4671 codec driver (currently hosted in for-2.6.33 branch). The driver has support for single ended inputs/outputs only, but the actual hardware supports full-differential mode as well. However, the chip doesn't have separate pins for the differential input/outputs, but instead the same pins used in single-ended mode can be reconfigured and reused. For example: LIN1 and RIN1 inputs (single-ended) can become IN1+/- (differential input), LIN2/RIN2 -> IN2+/-, etc. Similarly for outputs, LOUT1 and ROUT1 (single-ended) can become Mono Receiver RCP/RCN (BTL output), LOUT3/ROUT3 can become Mono Line Output LOP/LON (full-differential). Changing the function of a pin also modifies the audio map: - Mic Amp Left Mux allow to choose between LIN1, LIN2, LIN3 and LIN4, when pins are configured as single-ended inputs. However, if they are as differential inputs, then the options of Mic Amp Left Mux becomes only IN1+/- and IN3+/-. Similarly for Mic Amp Right Mux: IN2+/- and IN4+/-. - For outputs, single-ended stereo outputs LOUT1/ROUT1, LOUT2/ROUT2 and LOUT3/ROUT3 have separate analog mixers for each channel. But when the output pins work as RCP/RCN or LOP/LON, the output is mono and then each output has a single analog mixer. My concerns are: - When inputs/outputs are single-ended, then they have its own widgets, i.e.: SND_SOC_DAPM_INPUT("LIN1") SND_SOC_DAPM_INPUT("RIN1") But when they are switched to full-differential mode, they cannot have a separate widget anymore but a single widget instead: SND_SOC_DAPM_INPUT("IN1+/-") Then, it makes me think that input/output widgets cannot be reused in above case. I guess that having the three widgets at the same time is not a safe choice either. - Audio map changes depending on the mode of inputs or outputs, I don't know how the SoC codec driver could handle that. Any suggestion on what to do in above scenarios? Thanks, -Misa

2 1

[alsa-devel] Mail Delivery (failure)
by tsbogend＠alpha.franken.de 26 Nov '09

26 Nov '09

Binary message is available.

1 0

[PATCH 1/6] ASoC: add phycore-ac97 sound support
by Sascha Hauer 25 Nov '09

25 Nov '09

This patch adds sound support for Phytec PhyCORE / PhyCARD modules in AC97 mode. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> --- sound/soc/imx/Kconfig | 11 +++++- sound/soc/imx/Makefile | 2 + sound/soc/imx/phycore-ac97.c | 90 ++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 102 insertions(+), 1 deletions(-) create mode 100644 sound/soc/imx/phycore-ac97.c diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 5f006f0..e687d53 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -1,6 +1,6 @@ config SND_IMX_SOC tristate "SoC Audio for Freecale i.MX CPUs" - depends on ARCH_MXC && BROKEN + depends on ARCH_MXC select SND_PCM select FIQ select SND_SOC_AC97_BUS @@ -11,3 +11,12 @@ config SND_IMX_SOC config SND_MXC_SOC_SSI tristate +config SND_SOC_PHYCORE_AC97 + tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards" + depends on MACH_PCM043 || MACH_PCA100 + select SND_IMX_SOC + select SND_SOC_WM9712 + help + Say Y if you want to add support for SoC audio on Phytec phyCORE + and phyCARD boards in AC97 mode + diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile index d05cc95..9f8bb92 100644 --- a/sound/soc/imx/Makefile +++ b/sound/soc/imx/Makefile @@ -8,3 +8,5 @@ endif 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 diff --git a/sound/soc/imx/phycore-ac97.c b/sound/soc/imx/phycore-ac97.c new file mode 100644 index 0000000..a8307d5 --- /dev/null +++ b/sound/soc/imx/phycore-ac97.c @@ -0,0 +1,90 @@ +/* + * phycore-ac97.c -- SoC audio for imx_phycore in AC97 mode + * + * 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/wm9712.h" +#include "imx-ssi.h" + +static struct snd_soc_card imx_phycore; + +static struct snd_soc_ops imx_phycore_hifi_ops = { +}; + +static struct snd_soc_dai_link imx_phycore_dai_ac97[] = { + { + .name = "HiFi", + .stream_name = "HiFi", + .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI], + .ops = &imx_phycore_hifi_ops, + }, +}; + +static struct snd_soc_card imx_phycore = { + .name = "PhyCORE-audio", + .platform = &imx_soc_platform, + .dai_link = imx_phycore_dai_ac97, + .num_links = ARRAY_SIZE(imx_phycore_dai_ac97), +}; + +static struct snd_soc_device imx_phycore_snd_devdata = { + .card = &imx_phycore, + .codec_dev = &soc_codec_dev_wm9712, +}; + +static struct platform_device *imx_phycore_snd_device; + +static int __init imx_phycore_init(void) +{ + int ret; + + if (!machine_is_pcm043() && !machine_is_pca100()) + /* return happy. We might run on a totally different machine */ + return 0; + + imx_phycore_snd_device = platform_device_alloc("soc-audio", -1); + if (!imx_phycore_snd_device) + return -ENOMEM; + + imx_phycore_dai_ac97[0].cpu_dai = &imx_ssi_pcm_dai[0]; + + platform_set_drvdata(imx_phycore_snd_device, &imx_phycore_snd_devdata); + imx_phycore_snd_devdata.dev = &imx_phycore_snd_device->dev; + ret = platform_device_add(imx_phycore_snd_device); + + if (ret) { + printk(KERN_ERR "ASoC: Platform device allocation failed\n"); + platform_device_put(imx_phycore_snd_device); + } + + return ret; +} + +static void __exit imx_phycore_exit(void) +{ + platform_device_unregister(imx_phycore_snd_device); +} + +late_initcall(imx_phycore_init); +module_exit(imx_phycore_exit); + +MODULE_AUTHOR("Sascha Hauer <s.hauer(a)pengutronix.de>"); +MODULE_DESCRIPTION("PhyCORE ALSA SoC driver"); +MODULE_LICENSE("GPL"); -- 1.6.6 -- Pengutronix e.K. | | Industrial Linux Solutions | http://www.pengutronix.de/ | Peiner Str. 6-8, 31137 Hildesheim, Germany | Phone: +49-5121-206917-0 | Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-5555 |

1 0

[PATCH] ALSA: ASoc: Add regulator support to CS4270 codec driver
by Daniel Mack 25 Nov '09

25 Nov '09

Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Timur Tabi <timur(a)freescale.com> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> --- sound/soc/codecs/cs4270.c | 62 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 62 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 8069023..acbe961 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -28,6 +28,7 @@ #include <sound/initval.h> #include <linux/i2c.h> #include <linux/delay.h> +#include <linux/regulator/consumer.h> #include "cs4270.h" @@ -114,6 +115,10 @@ struct cs4270_private { unsigned int mode; /* The mode (I2S or left-justified) */ unsigned int slave_mode; unsigned int manual_mute; + + /* power domain regulators (optional) */ + struct regulator *va_reg; + bool va_reg_enabled; }; /** @@ -458,6 +463,42 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, } /** + * cs4270_set_bias_level - catches BIAS level changes + * @dai: the SOC DAI + * @level: the new BIAS level + * + * This function controls the voltage regulators to properly + * power up/down the voltage regulator domains + */ + +static int cs4270_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct cs4270_private *cs4270 = codec->private_data; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + /* Full power on */ + if (cs4270->va_reg && !cs4270->va_reg_enabled) { + regulator_enable(cs4270->va_reg); + cs4270->va_reg_enabled = true; + } + break; + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_OFF: + if (cs4270->va_reg && cs4270->va_reg_enabled) { + regulator_disable(cs4270->va_reg); + cs4270->va_reg_enabled = false; + } + break; + } + + codec->bias_level = level; + return 0; +} + +/** * cs4270_dai_mute - enable/disable the CS4270 external mute * @dai: the SOC DAI * @mute: 0 = disable mute, 1 = enable mute @@ -579,6 +620,7 @@ static int cs4270_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = cs4270_codec; + struct cs4270_private *cs4270 = codec->private_data; int ret; /* Connect the codec to the socdev. snd_soc_new_pcms() needs this. */ @@ -599,6 +641,13 @@ static int cs4270_probe(struct platform_device *pdev) goto error_free_pcms; } + /* get the power supply regulators (optional) */ + cs4270->va_reg = regulator_get(&pdev->dev, "va"); + if (IS_ERR(cs4270->va_reg)) + cs4270->va_reg = NULL; + + cs4270_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* And finally, register the socdev */ ret = snd_soc_init_card(socdev); if (ret < 0) { @@ -623,9 +672,13 @@ error_free_pcms: static int cs4270_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = cs4270_codec; + struct cs4270_private *cs4270 = codec->private_data; snd_soc_free_pcms(socdev); + regulator_put(cs4270->va_reg); + return 0; }; @@ -699,6 +752,8 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, codec->write = cs4270_i2c_write; codec->reg_cache = cs4270->reg_cache; codec->reg_cache_size = CS4270_NUMREGS; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = cs4270_set_bias_level; /* The I2C interface is set up, so pre-fill our register cache */ @@ -808,6 +863,8 @@ static int cs4270_i2c_suspend(struct i2c_client *client, pm_message_t mesg) struct cs4270_private *cs4270 = i2c_get_clientdata(client); struct snd_soc_codec *codec = &cs4270->codec; + cs4270_set_bias_level(codec, SND_SOC_BIAS_OFF); + return snd_soc_suspend_device(codec->dev); } @@ -816,6 +873,11 @@ static int cs4270_i2c_resume(struct i2c_client *client) struct cs4270_private *cs4270 = i2c_get_clientdata(client); struct snd_soc_codec *codec = &cs4270->codec; + cs4270_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) + cs4270_set_bias_level(codec, SND_SOC_BIAS_ON); + return snd_soc_resume_device(codec->dev); } -- 1.6.5.2

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[PATCH] ALSA: ASoc: Add regulator support to CS4270 codec driver
by Daniel Mack 25 Nov '09

25 Nov '09

Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Timur Tabi <timur(a)freescale.com> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> --- sound/soc/codecs/cs4270.c | 90 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 90 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 8069023..e79d30d 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -28,6 +28,7 @@ #include <sound/initval.h> #include <linux/i2c.h> #include <linux/delay.h> +#include <linux/regulator/consumer.h> #include "cs4270.h" @@ -114,6 +115,11 @@ struct cs4270_private { unsigned int mode; /* The mode (I2S or left-justified) */ unsigned int slave_mode; unsigned int manual_mute; + + /* power domain regulators (optional) */ + struct regulator *va_reg; + struct regulator *vd_reg; + struct regulator *vlc_reg; }; /** @@ -458,6 +464,56 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, } /** + * cs4270_set_bias_level - catches BIAS level changes + * @dai: the SOC DAI + * @level: the new BIAS level + * + * This function controls the voltage regulators to properly + * power up/down the voltage regulator domains + */ + +static int cs4270_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + int ret; + struct cs4270_private *cs4270 = codec->private_data; + + switch (level) { + case SND_SOC_BIAS_ON: + ret = regulator_enable(cs4270->vd_reg); + if (ret < 0) + return ret; + + ret = regulator_enable(cs4270->vlc_reg); + if (ret < 0) + return ret; + /* fall thru */ + case SND_SOC_BIAS_PREPARE: + ret = regulator_enable(cs4270->va_reg); + if (ret < 0) + return ret; + break; + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* OFF -> STANDBY */ + ret = regulator_enable(cs4270->va_reg); + if (ret < 0) + return ret; + } else + regulator_disable(cs4270->va_reg); + break; + case SND_SOC_BIAS_OFF: + regulator_disable(cs4270->va_reg); + regulator_disable(cs4270->vd_reg); + regulator_disable(cs4270->vlc_reg); + break; + } + + codec->bias_level = level; + return 0; +} + +/** * cs4270_dai_mute - enable/disable the CS4270 external mute * @dai: the SOC DAI * @mute: 0 = disable mute, 1 = enable mute @@ -579,6 +635,7 @@ static int cs4270_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = cs4270_codec; + struct cs4270_private *cs4270 = codec->private_data; int ret; /* Connect the codec to the socdev. snd_soc_new_pcms() needs this. */ @@ -599,6 +656,20 @@ static int cs4270_probe(struct platform_device *pdev) goto error_free_pcms; } + /* get the power supply regulators */ + cs4270->va_reg = regulator_get(&pdev->dev, "va"); + cs4270->vd_reg = regulator_get(&pdev->dev, "vd"); + cs4270->vlc_reg = regulator_get(&pdev->dev, "vlc"); + + if (IS_ERR(cs4270->va_reg) || + IS_ERR(cs4270->vd_reg) || + IS_ERR(cs4270->vlc_reg)) { + dev_err(codec->dev, "failed to get supplies %s\n", dev_name(codec->dev)); + goto error_free_pcms; + } + + cs4270_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* And finally, register the socdev */ ret = snd_soc_init_card(socdev); if (ret < 0) { @@ -609,6 +680,10 @@ static int cs4270_probe(struct platform_device *pdev) return 0; error_free_pcms: + regulator_put(cs4270->va_reg); + regulator_put(cs4270->vd_reg); + regulator_put(cs4270->vlc_reg); + snd_soc_free_pcms(socdev); return ret; @@ -623,9 +698,15 @@ error_free_pcms: static int cs4270_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = cs4270_codec; + struct cs4270_private *cs4270 = codec->private_data; snd_soc_free_pcms(socdev); + regulator_put(cs4270->va_reg); + regulator_put(cs4270->vd_reg); + regulator_put(cs4270->vlc_reg); + return 0; }; @@ -699,6 +780,8 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, codec->write = cs4270_i2c_write; codec->reg_cache = cs4270->reg_cache; codec->reg_cache_size = CS4270_NUMREGS; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = cs4270_set_bias_level; /* The I2C interface is set up, so pre-fill our register cache */ @@ -824,6 +907,8 @@ static int cs4270_soc_suspend(struct platform_device *pdev, pm_message_t mesg) struct snd_soc_codec *codec = cs4270_codec; int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL; + cs4270_set_bias_level(codec, SND_SOC_BIAS_OFF); + return snd_soc_write(codec, CS4270_PWRCTL, reg); } @@ -833,6 +918,11 @@ static int cs4270_soc_resume(struct platform_device *pdev) struct i2c_client *i2c_client = codec->control_data; int reg; + cs4270_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) + cs4270_set_bias_level(codec, SND_SOC_BIAS_ON); + /* In case the device was put to hard reset during sleep, we need to * wait 500ns here before any I2C communication. */ ndelay(500); -- 1.6.5.2

1 0

[PATCH] ALSA: ASoc: Add regulator support to CS4270 codec driver
by Daniel Mack 25 Nov '09

25 Nov '09

Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Timur Tabi <timur(a)freescale.com> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> --- sound/soc/codecs/cs4270.c | 114 ++++++++++++++++++++++++++++++++++++++++++++- 1 files changed, 112 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 8069023..34cd97d 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -28,6 +28,7 @@ #include <sound/initval.h> #include <linux/i2c.h> #include <linux/delay.h> +#include <linux/regulator/consumer.h> #include "cs4270.h" @@ -114,6 +115,12 @@ struct cs4270_private { unsigned int mode; /* The mode (I2S or left-justified) */ unsigned int slave_mode; unsigned int manual_mute; + + /* power domain regulators */ + struct regulator *va_reg; + struct regulator *vd_reg; + struct regulator *vlc_reg; + bool va_reg_enabled; }; /** @@ -458,6 +465,50 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, } /** + * cs4270_set_bias_level - catches BIAS level changes + * @dai: the SOC DAI + * @level: the new BIAS level + * + * This function controls the voltage regulators to properly + * power up/down the voltage regulator domains + */ + +static int cs4270_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + int ret; + struct cs4270_private *cs4270 = codec->private_data; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + if (!cs4270->va_reg_enabled) { + ret = regulator_enable(cs4270->va_reg); + if (ret < 0) + return ret; + + cs4270->va_reg_enabled = true; + } + + break; + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_OFF: + if (cs4270->va_reg_enabled) { + ret = regulator_disable(cs4270->va_reg); + if (ret < 0) + return ret; + + cs4270->va_reg_enabled = false; + } + + break; + } + + codec->bias_level = level; + return 0; +} + +/** * cs4270_dai_mute - enable/disable the CS4270 external mute * @dai: the SOC DAI * @mute: 0 = disable mute, 1 = enable mute @@ -579,6 +630,7 @@ static int cs4270_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = cs4270_codec; + struct cs4270_private *cs4270 = codec->private_data; int ret; /* Connect the codec to the socdev. snd_soc_new_pcms() needs this. */ @@ -599,6 +651,25 @@ static int cs4270_probe(struct platform_device *pdev) goto error_free_pcms; } + /* get the power supply regulators */ + cs4270->va_reg = regulator_get(codec->dev, "va"); + cs4270->vd_reg = regulator_get(codec->dev, "vd"); + cs4270->vlc_reg = regulator_get(codec->dev, "vlc"); + + if (IS_ERR(cs4270->va_reg) || + IS_ERR(cs4270->vd_reg) || + IS_ERR(cs4270->vlc_reg)) { + dev_err(codec->dev, + "failed to get supplies for device %s\n", + dev_name(codec->dev)); + goto error_free_pcms; + } + + regulator_enable(cs4270->vd_reg); + regulator_enable(cs4270->vlc_reg); + + cs4270_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* And finally, register the socdev */ ret = snd_soc_init_card(socdev); if (ret < 0) { @@ -609,6 +680,10 @@ static int cs4270_probe(struct platform_device *pdev) return 0; error_free_pcms: + regulator_put(cs4270->va_reg); + regulator_put(cs4270->vd_reg); + regulator_put(cs4270->vlc_reg); + snd_soc_free_pcms(socdev); return ret; @@ -623,9 +698,21 @@ error_free_pcms: static int cs4270_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = cs4270_codec; + struct cs4270_private *cs4270 = codec->private_data; snd_soc_free_pcms(socdev); + if (cs4270->va_reg_enabled) + regulator_disable(cs4270->va_reg); + + regulator_disable(cs4270->vd_reg); + regulator_disable(cs4270->vlc_reg); + + regulator_put(cs4270->va_reg); + regulator_put(cs4270->vd_reg); + regulator_put(cs4270->vlc_reg); + return 0; }; @@ -699,6 +786,8 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, codec->write = cs4270_i2c_write; codec->reg_cache = cs4270->reg_cache; codec->reg_cache_size = CS4270_NUMREGS; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = cs4270_set_bias_level; /* The I2C interface is set up, so pre-fill our register cache */ @@ -822,17 +911,38 @@ static int cs4270_i2c_resume(struct i2c_client *client) static int cs4270_soc_suspend(struct platform_device *pdev, pm_message_t mesg) { struct snd_soc_codec *codec = cs4270_codec; - int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL; + struct cs4270_private *cs4270 = codec->private_data; + int reg, ret; - return snd_soc_write(codec, CS4270_PWRCTL, reg); + reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL; + if (reg < 0) + return reg; + + ret = snd_soc_write(codec, CS4270_PWRCTL, reg); + if (ret < 0) + return ret; + + ret = cs4270_set_bias_level(codec, SND_SOC_BIAS_OFF); + if (ret < 0) + return ret; + + regulator_disable(cs4270->vd_reg); + regulator_disable(cs4270->vlc_reg); + + return 0; } static int cs4270_soc_resume(struct platform_device *pdev) { struct snd_soc_codec *codec = cs4270_codec; + struct cs4270_private *cs4270 = codec->private_data; struct i2c_client *i2c_client = codec->control_data; int reg; + regulator_enable(cs4270->vd_reg); + regulator_enable(cs4270->vlc_reg); + cs4270_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* In case the device was put to hard reset during sleep, we need to * wait 500ns here before any I2C communication. */ ndelay(500); -- 1.6.5.2

1 0

[PATCH] ALSA: ASoc: Add regulator support to CS4270 codec driver
by Daniel Mack 25 Nov '09

25 Nov '09

Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Timur Tabi <timur(a)freescale.com> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> --- sound/soc/codecs/cs4270.c | 114 ++++++++++++++++++++++++++++++++++++++++++++- 1 files changed, 112 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index ffe122d..279a644 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -28,6 +28,7 @@ #include <sound/initval.h> #include <linux/i2c.h> #include <linux/delay.h> +#include <linux/regulator/consumer.h> #include "cs4270.h" @@ -114,6 +115,12 @@ struct cs4270_private { unsigned int mode; /* The mode (I2S or left-justified) */ unsigned int slave_mode; unsigned int manual_mute; + + /* power domain regulators */ + struct regulator *va_reg; + struct regulator *vd_reg; + struct regulator *vlc_reg; + bool va_reg_enabled; }; /** @@ -458,6 +465,50 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, } /** + * cs4270_set_bias_level - catches BIAS level changes + * @dai: the SOC DAI + * @level: the new BIAS level + * + * This function controls the voltage regulators to properly + * power up/down the voltage regulator domains + */ + +static int cs4270_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + int ret; + struct cs4270_private *cs4270 = codec->private_data; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + if (!cs4270->va_reg_enabled) { + ret = regulator_enable(cs4270->va_reg); + if (ret < 0) + return ret; + + cs4270->va_reg_enabled = true; + } + + break; + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_OFF: + if (cs4270->va_reg_enabled) { + ret = regulator_disable(cs4270->va_reg); + if (ret < 0) + return ret; + + cs4270->va_reg_enabled = false; + } + + break; + } + + codec->bias_level = level; + return 0; +} + +/** * cs4270_dai_mute - enable/disable the CS4270 external mute * @dai: the SOC DAI * @mute: 0 = disable mute, 1 = enable mute @@ -579,6 +630,7 @@ static int cs4270_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = cs4270_codec; + struct cs4270_private *cs4270 = codec->private_data; int ret; /* Connect the codec to the socdev. snd_soc_new_pcms() needs this. */ @@ -599,9 +651,32 @@ static int cs4270_probe(struct platform_device *pdev) goto error_free_pcms; } + /* get the power supply regulators */ + cs4270->va_reg = regulator_get(codec->dev, "va"); + cs4270->vd_reg = regulator_get(codec->dev, "vd"); + cs4270->vlc_reg = regulator_get(codec->dev, "vlc"); + + if (IS_ERR(cs4270->va_reg) || + IS_ERR(cs4270->vd_reg) || + IS_ERR(cs4270->vlc_reg)) { + dev_err(codec->dev, + "failed to get supplies for device %s\n", + dev_name(codec->dev)); + goto error_free_pcms; + } + + regulator_enable(cs4270->vd_reg); + regulator_enable(cs4270->vlc_reg); + + cs4270_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; error_free_pcms: + regulator_put(cs4270->va_reg); + regulator_put(cs4270->vd_reg); + regulator_put(cs4270->vlc_reg); + snd_soc_free_pcms(socdev); return ret; @@ -616,9 +691,21 @@ error_free_pcms: static int cs4270_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = cs4270_codec; + struct cs4270_private *cs4270 = codec->private_data; snd_soc_free_pcms(socdev); + if (cs4270->va_reg_enabled) + regulator_disable(cs4270->va_reg); + + regulator_disable(cs4270->vd_reg); + regulator_disable(cs4270->vlc_reg); + + regulator_put(cs4270->va_reg); + regulator_put(cs4270->vd_reg); + regulator_put(cs4270->vlc_reg); + return 0; }; @@ -692,6 +779,8 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, codec->write = cs4270_i2c_write; codec->reg_cache = cs4270->reg_cache; codec->reg_cache_size = CS4270_NUMREGS; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = cs4270_set_bias_level; /* The I2C interface is set up, so pre-fill our register cache */ @@ -799,17 +888,38 @@ MODULE_DEVICE_TABLE(i2c, cs4270_id); static int cs4270_soc_suspend(struct platform_device *pdev, pm_message_t mesg) { struct snd_soc_codec *codec = cs4270_codec; - int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL; + struct cs4270_private *cs4270 = codec->private_data; + int reg, ret; - return snd_soc_write(codec, CS4270_PWRCTL, reg); + reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL; + if (reg < 0) + return reg; + + ret = snd_soc_write(codec, CS4270_PWRCTL, reg); + if (ret < 0) + return ret; + + ret = cs4270_set_bias_level(codec, SND_SOC_BIAS_OFF); + if (ret < 0) + return ret; + + regulator_disable(cs4270->vd_reg); + regulator_disable(cs4270->vlc_reg); + + return 0; } static int cs4270_soc_resume(struct platform_device *pdev) { struct snd_soc_codec *codec = cs4270_codec; + struct cs4270_private *cs4270 = codec->private_data; struct i2c_client *i2c_client = codec->control_data; int reg; + regulator_enable(cs4270->vd_reg); + regulator_enable(cs4270->vlc_reg); + cs4270_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* In case the device was put to hard reset during sleep, we need to * wait 500ns here before any I2C communication. */ ndelay(500); -- 1.6.5.2

1 0

[alsa-devel] [PATCH 2/4] ASoC: DaVinci: Voice Codec Interface
by miguel.aguilar＠ridgerun.com 25 Nov '09

25 Nov '09

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> 1) Adds an interface needed by the voice codec CQ0093. 2) Add an option to select internal or external codec in the DM365. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- sound/soc/davinci/Kconfig | 26 +++- sound/soc/davinci/Makefile | 2 + sound/soc/davinci/davinci-vcif.c | 363 ++++++++++++++++++++++++++++++++++++++ sound/soc/davinci/davinci-vcif.h | 63 +++++++ 4 files changed, 452 insertions(+), 2 deletions(-) create mode 100644 sound/soc/davinci/davinci-vcif.c create mode 100644 sound/soc/davinci/davinci-vcif.h diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 047ee39..051a10e 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -9,18 +9,40 @@ config SND_DAVINCI_SOC config SND_DAVINCI_SOC_I2S tristate +config SND_DAVINCI_SOC_VCIF + tristate + config SND_DAVINCI_SOC_MCASP tristate config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" depends on SND_DAVINCI_SOC - depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM + depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM select SND_DAVINCI_SOC_I2S select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on TI - DaVinci DM6446 or DM355 EVM platforms. + DaVinci DM6446, DM355 or DM365 EVM platforms. + +choice + prompt "DM365 codec select" + depends on SND_DAVINCI_SOC_EVM + depends on MACH_DAVINCI_DM365_EVM + default SND_DM365_EXTERNAL_CODEC + +config SND_DM365_INTERNAL_CODEC + bool "CQ93VC" + select SND_DAVINCI_SOC_VCIF + select SND_SOC_CQ0093VC + help + Say Y if you want to add support for SoC On-chip voice codec + +config SND_DM365_EXTERNAL_CODEC + bool "AIC3101" + help + Say Y if you want to add support for AIC3101 audio codec +endchoice config SND_DM6467_SOC_EVM tristate "SoC Audio support for DaVinci DM6467 EVM" diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index a6939d7..a93679d 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -2,10 +2,12 @@ snd-soc-davinci-objs := davinci-pcm.o snd-soc-davinci-i2s-objs := davinci-i2s.o snd-soc-davinci-mcasp-objs:= davinci-mcasp.o +snd-soc-davinci-vcif-objs:= davinci-vcif.o obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o +obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o # DAVINCI Machine Support snd-soc-evm-objs := davinci-evm.o diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c new file mode 100644 index 0000000..83191f1 --- /dev/null +++ b/sound/soc/davinci/davinci-vcif.c @@ -0,0 +1,363 @@ +/* + * ALSA SoC Voice Codec Interface for TI DAVINCI processor + * + * Copyright (C) 2009 Texas Instruments. + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * Initial code: Hui Geng + * + * 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/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "davinci-pcm.h" +#include "davinci-i2s.h" +#include "davinci-vcif.h" + +#define MOD_REG_BIT(val, mask, set) do { \ + if (set) { \ + val |= mask; \ + } else { \ + val &= ~mask; \ + } \ +} while (0) + +static struct davinci_pcm_dma_params davinci_vcif_pcm_out = { + .name = "VCIF PCM Stereo out", +}; + +static struct davinci_pcm_dma_params davinci_vcif_pcm_in = { + .name = "VCIF PCM Stereo in", +}; + +struct davinci_vcif_dev { + void __iomem *base; + resource_size_t pbase; + size_t base_size; + int mode; + u32 pcr; + struct clk *clk; + struct davinci_pcm_dma_params *dma_params[2]; +}; + +static inline void davinci_vcif_write_reg(struct davinci_vcif_dev *dev, + int reg, u32 val) +{ + __raw_writel(val, dev->base + reg); +} + +static inline u32 davinci_vcif_read_reg(struct davinci_vcif_dev *dev, int reg) +{ + return __raw_readl(dev->base + reg); +} + +static void davinci_vcif_start(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_vcif_dev *dev = rtd->dai->cpu_dai->private_data; + u32 w; + + /* Start the sample generator and enable transmitter/receiver */ + w = davinci_vcif_read_reg(dev, DAVINCI_VCIF_CTRL_REG); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + MOD_REG_BIT(w, DAVINCI_VCIF_CTRL_RSTDAC, 1); + else + MOD_REG_BIT(w, DAVINCI_VCIF_CTRL_RSTADC, 1); + + davinci_vcif_write_reg(dev, DAVINCI_VCIF_CTRL_REG, w); +} + +static void davinci_vcif_stop(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_vcif_dev *dev = rtd->dai->cpu_dai->private_data; + u32 w; + + /* Reset transmitter/receiver and sample rate/frame sync generators */ + w = davinci_vcif_read_reg(dev, DAVINCI_VCIF_CTRL_REG); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + MOD_REG_BIT(w, DAVINCI_VCIF_CTRL_RSTDAC, 0); + else + MOD_REG_BIT(w, DAVINCI_VCIF_CTRL_RSTADC, 0); + + davinci_vcif_write_reg(dev, DAVINCI_VCIF_CTRL_REG, w); +} + +static int davinci_vcif_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct davinci_vcif_dev *dev = cpu_dai->private_data; + + cpu_dai->dma_data = dev->dma_params[substream->stream]; + return 0; +} + +static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct davinci_pcm_dma_params *dma_params = dai->dma_data; + struct davinci_vcif_dev *dev = dai->private_data; + u32 w; + + /* Restart the codec before setup */ + davinci_vcif_stop(substream); + davinci_vcif_start(substream); + + /* General line settings */ + davinci_vcif_write_reg(dev, + DAVINCI_VCIF_CTRL_REG, DAVINCI_VCIF_CTRL_MASK); + + davinci_vcif_write_reg(dev, + DAVINCI_VCIF_INTCLR_REG, DAVINCI_VCIF_INT_MASK); + + davinci_vcif_write_reg(dev, + DAVINCI_VCIF_INTEN_REG, DAVINCI_VCIF_INT_MASK); + + w = davinci_vcif_read_reg(dev, DAVINCI_VCIF_CTRL_REG); + + /* Determine xfer data type */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + dma_params->data_type = 0; + + MOD_REG_BIT(w, DAVINCI_VCIF_CTRL_RD_BITS_8 | + DAVINCI_VCIF_CTRL_RD_UNSIGNED | + DAVINCI_VCIF_CTRL_WD_BITS_8 | + DAVINCI_VCIF_CTRL_WD_UNSIGNED, 1); + break; + case SNDRV_PCM_FORMAT_S8: + dma_params->data_type = 1; + + MOD_REG_BIT(w, DAVINCI_VCIF_CTRL_RD_BITS_8 | + DAVINCI_VCIF_CTRL_WD_BITS_8, 1); + + MOD_REG_BIT(w, DAVINCI_VCIF_CTRL_RD_UNSIGNED | + DAVINCI_VCIF_CTRL_WD_UNSIGNED, 0); + break; + case SNDRV_PCM_FORMAT_S16_LE: + dma_params->data_type = 2; + + MOD_REG_BIT(w, DAVINCI_VCIF_CTRL_RD_BITS_8 | + DAVINCI_VCIF_CTRL_RD_UNSIGNED | + DAVINCI_VCIF_CTRL_WD_BITS_8 | + DAVINCI_VCIF_CTRL_WD_UNSIGNED, 0); + break; + default: + printk(KERN_WARNING "davinci-vcif: unsupported PCM format"); + return -EINVAL; + } + + dma_params->acnt = dma_params->data_type; + + davinci_vcif_write_reg(dev, DAVINCI_VCIF_CTRL_REG, w); + + return 0; +} + +static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + davinci_vcif_start(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_vcif_stop(substream); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +#define DAVINCI_VCIF_RATES SNDRV_PCM_RATE_8000_48000 + +static struct snd_soc_dai_ops davinci_vcif_dai_ops = { + .startup = davinci_vcif_startup, + .trigger = davinci_vcif_trigger, + .hw_params = davinci_vcif_hw_params, +}; + +struct snd_soc_dai davinci_vcif_dai = { + .name = "davinci-vcif", + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = DAVINCI_VCIF_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = DAVINCI_VCIF_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &davinci_vcif_dai_ops, + +}; +EXPORT_SYMBOL_GPL(davinci_vcif_dai); + +static int davinci_vcif_probe(struct platform_device *pdev) +{ + struct davinci_vcif_dev *dev; + struct resource *res, *mem; + int ret; + + dev = kzalloc(sizeof(struct davinci_vcif_dev), GFP_KERNEL); + if (!dev) { + dev_dbg(&pdev->dev, "could not allocate memory for private data\n"); + return -ENOMEM; + } + + dev->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(dev->clk)) { + dev_dbg(&pdev->dev, "%s: could not get the clock for voice codec\n", + pdev->name); + ret = -ENODEV; + goto fail1; + } + clk_enable(dev->clk); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "no mem resource\n"); + ret = -ENODEV; + goto fail2; + } + + dev->pbase = res->start; + dev->base_size = resource_size(res); + + mem = request_mem_region(dev->pbase, dev->base_size, pdev->name); + if (!mem) { + dev_err(&pdev->dev, "VCIF region already claimed\n"); + ret = -EBUSY; + goto fail2; + } + + dev->base = ioremap(dev->pbase, dev->base_size); + if (!dev->base) { + dev_err(&pdev->dev,"%s: can't ioremap mem resource.\n", pdev->name); + ret = -ENOMEM; + goto fail3; + } + + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &davinci_vcif_pcm_out; + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->dma_addr = + (dma_addr_t)(io_v2p(dev->base) + DAVINCI_VCIF_WFIFO_REG); + + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &davinci_vcif_pcm_in; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->dma_addr = + (dma_addr_t)(io_v2p(dev->base) + DAVINCI_VCIF_RFIFO_REG); + + /* first TX, then RX */ + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(&pdev->dev, "%s: no DMA resource\n", pdev->name); + ret = -ENXIO; + goto fail4; + } + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = res->start; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(&pdev->dev, "%s: no DMA resource\n", pdev->name); + ret = -ENXIO; + goto fail4; + } + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = res->start; + + davinci_vcif_dai.private_data = dev; + ret = snd_soc_register_dai(&davinci_vcif_dai); + if (ret != 0) + goto fail4; + + return 0; + +fail4: + iounmap(dev->base); +fail3: + release_mem_region(dev->pbase, dev->base_size); +fail2: + clk_disable(dev->clk); + clk_put(dev->clk); + dev->clk = NULL; +fail1: + kfree(dev); + + return ret; +} + +static int davinci_vcif_remove(struct platform_device *pdev) +{ + struct davinci_vcif_dev *dev = davinci_vcif_dai.private_data; + + iounmap(dev->base); + release_mem_region(dev->pbase, dev->base_size); + + clk_disable(dev->clk); + clk_put(dev->clk); + dev->clk = NULL; + + kfree(dev); + + snd_soc_unregister_dai(&davinci_vcif_dai); + + return 0; +} + +static struct platform_driver davinci_vcif_driver = { + .probe = davinci_vcif_probe, + .remove = davinci_vcif_remove, + .driver = { + .name = "voice_codec", + .owner = THIS_MODULE, + }, +}; + +static int __init davinci_vcif_init(void) +{ + return platform_driver_probe(&davinci_vcif_driver, davinci_vcif_probe); +} +module_init(davinci_vcif_init); + +static void __exit davinci_vcif_exit(void) +{ + platform_driver_unregister(&davinci_vcif_driver); +} +module_exit(davinci_vcif_exit); + +MODULE_AUTHOR("Miguel Aguilar"); +MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC Voice Codec Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/davinci/davinci-vcif.h b/sound/soc/davinci/davinci-vcif.h new file mode 100644 index 0000000..30e441d --- /dev/null +++ b/sound/soc/davinci/davinci-vcif.h @@ -0,0 +1,63 @@ +/* + * ALSA SoC Voice Codec Interface for TI DAVINCI processor + * + * Copyright (C) 2009 Texas Instruments. + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * Initial code: Hui Geng + * + * 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 + */ + +#ifndef _DAVINCI_VCIF_H +#define _DAVINCI_VCIF_H + +#define DAVINCI_VCIF_PID_REG 0x00 +#define DAVINCI_VCIF_CTRL_REG 0x04 +#define DAVINCI_VCIF_INTEN_REG 0x08 +#define DAVINCI_VCIF_INTSTATUS_REG 0x0c +#define DAVINCI_VCIF_INTCLR_REG 0x10 +#define DAVINCI_VCIF_EMUL_CTRL_REG 0x14 +#define DAVINCI_VCIF_RFIFO_REG 0x20 +#define DAVINCI_VCIF_WFIFO_REG 0x24 +#define DAVINCI_VCIF_FIFOSTAT_REG 0x28 +#define DAVINCI_VCIF_TST_CTRL_REG 0x2C + +#define DAVINCI_VCIF_CTRL_MASK 0x5500 +#define DAVINCI_VCIF_CTRL_RSTADC (1 << 0) +#define DAVINCI_VCIF_CTRL_RSTDAC (1 << 1) +#define DAVINCI_VCIF_CTRL_RD_BITS_8 (1 << 4) +#define DAVINCI_VCIF_CTRL_RD_UNSIGNED (1 << 5) +#define DAVINCI_VCIF_CTRL_WD_BITS_8 (1 << 6) +#define DAVINCI_VCIF_CTRL_WD_UNSIGNED (1 << 7) +#define DAVINCI_VCIF_CTRL_RFIFOEN (1 << 8) +#define DAVINCI_VCIF_CTRL_RFIFOCL (1 << 9) +#define DAVINCI_VCIF_CTRL_RFIFOMD_WORD_1 (1 << 10) +#define DAVINCI_VCIF_CTRL_WFIFOEN (1 << 12) +#define DAVINCI_VCIF_CTRL_WFIFOCL (1 << 13) +#define DAVINCI_VCIF_CTRL_WFIFOMD_WORD_1 (1 << 14) + +#define DAVINCI_VCIF_INT_MASK 0x3F +#define DAVINCI_VCIF_INT_RDRDY_MASK (1 << 0) +#define DAVINCI_VCIF_INT_RERROVF_MASK (1 << 1) +#define DAVINCI_VCIF_INT_RERRUDR_MASK (1 << 2) +#define DAVINCI_VCIF_INT_WDREQ_MASK (1 << 3) +#define DAVINCI_VCIF_INT_WERROVF_MASK (1 << 4) +#define DAVINCI_VCIF_INT_WERRUDR_MASK (1 << 5) + +extern struct snd_soc_dai davinci_vcif_dai; + +#endif -- 1.6.0.4

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[alsa-devel] [PATCH] ALSA: intel8x0: Mute External Amplifier by default for Gateway 4525GZ
by Daniel Chen 25 Nov '09

25 Nov '09

Hi, Attached please find a patch against current sound-2.6 HEAD setting the inv_eapd quirk for the Gateway 4525GZ. Thanks, Dan

2 1

[alsa-devel] USB patch failures alsa-driver git
by Eliot Blennerhassett 24 Nov '09

24 Nov '09

Greetings, trying to build alsa-driver, I get the following patch failures??? Maybe recent alsa-kernel usb changes have broken the patches in alsa-driver? (for now, I get around the problem by copying the original files instead of patching) uname -a Linux 2.6.28-16-generic #55-Ubuntu SMP Tue Oct 20 19:48:24 UTC 2009 i686 GNU/Linux copying file alsa-kernel/usb/usbmidi.c patching file usbmidi.c Hunk #2 succeeded at 245 (offset 8 lines). Hunk #3 FAILED at 269. Hunk #4 FAILED at 392. Hunk #5 succeeded at 1699 (offset 97 lines). Hunk #6 succeeded at 2070 (offset 102 lines). 2 out of 6 hunks FAILED -- saving rejects to file usbmidi.c.rej make[1]: *** [usbmidi.c] Error 1 make[1]: Leaving directory `/home/eliot/src/alsagit/alsa-driver/usb' make: *** [modules] Error 2 copying file alsa-kernel/usb/usx2y/usX2Yhwdep.c patching file usX2Yhwdep.c Hunk #5 FAILED at 196. Hunk #6 succeeded at 252 with fuzz 2. 1 out of 7 hunks FAILED -- saving rejects to file usX2Yhwdep.c.rej make[2]: *** [usX2Yhwdep.c] Error 1 make[2]: Leaving directory `/home/eliot/src/alsagit/alsa-driver/usb/usx2y' make[1]: *** [prepare] Error 1 make[1]: Leaving directory `/home/eliot/src/alsagit/alsa-driver/usb' make: *** [modules] Error 2 copying file alsa-kernel/usb/usx2y/usbusx2y.c patching file usbusx2y.c Hunk #4 succeeded at 268 with fuzz 1. Hunk #5 FAILED at 281. Hunk #7 succeeded at 391 with fuzz 2 (offset -1 lines). Hunk #8 succeeded at 415 (offset -2 lines). Hunk #9 succeeded at 441 (offset -1 lines). Hunk #10 succeeded at 462 (offset -1 lines). Hunk #11 succeeded at 522 (offset -2 lines). Hunk #12 succeeded at 542 (offset -2 lines). copying file alsa-kernel/usb/usx2y/usbusx2yaudio.c patching file usbusx2yaudio.c Hunk #3 succeeded at 316 (offset 1 line). Hunk #4 succeeded at 349 (offset 1 line). Hunk #5 succeeded at 365 (offset 1 line). Hunk #6 succeeded at 391 (offset 1 line). Hunk #7 succeeded at 407 (offset 1 line). Hunk #8 succeeded at 524 (offset 1 line). Hunk #9 succeeded at 547 (offset 1 line). Hunk #10 succeeded at 696 (offset 1 line). Hunk #11 FAILED at 1059. 0

2 1

[alsa-devel] ASoC update for 2.6.32
by Mark Brown 24 Nov '09

24 Nov '09

The following changes since commit bab0212467e58929470ae3ae32515f17e30c3926: Troy Kisky (1): ASoC: tlv320aic23 fix rate selection are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 It might be a little late to get this one in but it's a nasty bug. Mark Brown (1): ASoC: Fix suspend with active audio streams sound/soc/soc-dapm.c | 20 +++++++++++++++++--- 1 files changed, 17 insertions(+), 3 deletions(-)

2 1

[alsa-devel] Patch: Make Dell Vostro 1015n mic and speaker switching work
by Einar Rünkaru 24 Nov '09

24 Nov '09

Hi. Dell Vostro 1015n uses Conexant CX20583-10Z (0x14f1:5067). Patch is based on "olpc-xo-1_5" branch. The difference is that Dell uses digital mic. Tested on Ubuntu 9.04 Einar

2 3

[alsa-devel] ASoC update for 2.6.33
by Mark Brown 23 Nov '09

23 Nov '09

The following changes since commit dcdec639ad74f20427df82284f98b68f6ac7c0ae: Mark Brown (1): Merge branch 'ads117x' of git://git.kernel.org/.../broonie/sound-2.6 into for-2.6.33 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Mark Brown (2): ASoC: Fix suspend with active audio streams Merge branch 'for-2.6.32' into for-2.6.33 sound/soc/soc-dapm.c | 20 +++++++++++++++++--- 1 files changed, 17 insertions(+), 3 deletions(-)

1 0

[alsa-devel] [PATCH 6/8] Audio driver of Always Innovating OMAP3-based Touch Book
by Gregoire Gentil 23 Nov '09

23 Nov '09

Signed-off-by: Gregoire Gentil <gregoire(a)gentil.com> --- sound/soc/omap/Kconfig | 8 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/omap3touchbook.c | 211 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 221 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/omap3touchbook.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 653a362..c590252 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -83,6 +83,14 @@ config SND_OMAP_SOC_OMAP3_PANDORA help Say Y if you want to add support for SoC audio on the OMAP3 Pandora. +config SND_OMAP_SOC_OMAP3_TOUCHBOOK + tristate "SoC Audio support for OMAP3 Touch Book" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_TOUCHBOOK + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the OMAP3 Touch Book. + config SND_OMAP_SOC_OMAP3_BEAGLE tristate "SoC Audio support for OMAP3 Beagle" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_BEAGLE diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 02d6947..f360b82 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -15,6 +15,7 @@ snd-soc-omap3evm-objs := omap3evm.o snd-soc-sdp3430-objs := sdp3430.o snd-soc-omap3pandora-objs := omap3pandora.o snd-soc-omap3beagle-objs := omap3beagle.o +snd-soc-omap3touchbook-objs := omap3touchbook.o snd-soc-zoom2-objs := zoom2.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o @@ -26,4 +27,5 @@ obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(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_OMAP3_TOUCHBOOK) += snd-soc-omap3touchbook.o obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o diff --git a/sound/soc/omap/omap3touchbook.c b/sound/soc/omap/omap3touchbook.c new file mode 100644 index 0000000..584f07d --- /dev/null +++ b/sound/soc/omap/omap3touchbook.c @@ -0,0 +1,211 @@ +/* + * omap3touchbook.c -- SoC audio for OMAP3 Touch Book + * + * Author: Gregoire Gentil <gregoire(a)gentil.com> + * + * Derived from sound/soc/omap/omap3beagle.c + * + * 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/irq.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <plat/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +#define TB_HEADPHONE_GPIO 56 +#define TB_HEADPHONE_IRQ OMAP_GPIO_IRQ(TB_HEADPHONE_GPIO) + +static int omap3touchbook_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int fmt; + int ret; + + switch (params_channels(params)) { + case 2: /* Stereo I2S mode */ + fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + break; + case 4: /* Four channel TDM mode */ + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM; + break; + default: + return -EINVAL; + } + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops omap3touchbook_ops = { + .hw_params = omap3touchbook_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link omap3touchbook_dai = { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], + .ops = &omap3touchbook_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_omap3touchbook = { + .name = "omap3touchbook", + .platform = &omap_soc_platform, + .dai_link = &omap3touchbook_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device omap3touchbook_snd_devdata = { + .card = &snd_soc_omap3touchbook, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *omap3touchbook_snd_device; + +static void jack_work_func(struct work_struct *wq) +{ + struct snd_soc_device *socdev = platform_get_drvdata( + omap3touchbook_snd_device); + struct snd_soc_codec *codec = socdev->card->codec; + + snd_soc_dapm_sync(codec); +} +DECLARE_WORK(jack_work, jack_work_func); + +static irqreturn_t touchbook_headphone_event(int irq, void *snd) +{ + int status = gpio_get_value(TB_HEADPHONE_GPIO); + struct snd_soc_device *socdev = platform_get_drvdata( + omap3touchbook_snd_device); + struct snd_soc_codec *codec = socdev->card->codec; + + if (status) { + snd_soc_dapm_disable_pin(codec, "HFL"); + snd_soc_dapm_disable_pin(codec, "HFR"); + } else { + snd_soc_dapm_enable_pin(codec, "HFL"); + snd_soc_dapm_enable_pin(codec, "HFR"); + } + + schedule_work(&jack_work); + return IRQ_HANDLED; +} + +static int __init omap3touchbook_soc_init(void) +{ + int ret; + + if (!machine_is_touchbook()) { + pr_debug("Not OMAP3 Touch Book!\n"); + return -ENODEV; + } + pr_info("OMAP3 Touch Book SoC init\n"); + + omap3touchbook_snd_device = platform_device_alloc("soc-audio", -1); + if (!omap3touchbook_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(omap3touchbook_snd_device, + &omap3touchbook_snd_devdata); + omap3touchbook_snd_devdata.dev = &omap3touchbook_snd_device->dev; + *(unsigned int *)omap3touchbook_dai.cpu_dai->private_data = 1; + + /* Touch Book -- headphone jack sensor */ + omap_set_gpio_debounce(TB_HEADPHONE_GPIO, 1); + omap_set_gpio_debounce_time(TB_HEADPHONE_GPIO, 0xff); + + ret = request_irq(TB_HEADPHONE_IRQ, touchbook_headphone_event, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING, + "touchbook_headphone", + omap3touchbook_snd_device); + if (ret < 0) + goto err1; + + ret = platform_device_add(omap3touchbook_snd_device); + if (ret) + goto err1; + + /* Detect headphone status */ + touchbook_headphone_event(0, omap3touchbook_snd_device); + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(omap3touchbook_snd_device); + + return ret; +} + +static void __exit omap3touchbook_soc_exit(void) +{ + platform_device_unregister(omap3touchbook_snd_device); +} + +module_init(omap3touchbook_soc_init); +module_exit(omap3touchbook_soc_exit); + +MODULE_AUTHOR("Gregoire Gentil <gregoire(a)gentil.com>"); +MODULE_DESCRIPTION("ALSA SoC OMAP3 Touch Book"); +MODULE_LICENSE("GPL"); -- 1.6.0.4

2 1

[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 23 Nov '09

23 Nov '09

The following changes since commit 41b51dd47ea7c406a8d49b97804e8acec9dadaed: Mark Brown (1): Merge branch 'for-2.6.32' into for-2.6.33 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Enric Balletbò i Serra (1): ASoC: Add support for IGEP v2 Graeme Gregory (1): ASoC: ADS117x ADC driver Mark Brown (2): ASoC: Update ads117x to current APIs Merge branch 'ads117x' of git://git.kernel.org/.../broonie/sound-2.6 into for-2.6.33 Troy Kisky (4): ASoC: DaVinci: i2s, reduce underruns by combining into 1 element ASoC: DaVinci: pcm, rename variables in prep for ping/pong ASoC: DaVinci: pcm, fix underrun by using sram ASoC: DaVinci: use edma_pause, edma_resume arch/arm/mach-davinci/include/mach/asp.h | 8 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ads117x.c | 123 +++++++ sound/soc/codecs/ads117x.h | 13 + sound/soc/davinci/davinci-i2s.c | 79 ++++- sound/soc/davinci/davinci-pcm.c | 551 ++++++++++++++++++++++++++---- sound/soc/davinci/davinci-pcm.h | 1 + sound/soc/omap/Kconfig | 7 + sound/soc/omap/Makefile | 2 + sound/soc/omap/igep0020.c | 148 ++++++++ 11 files changed, 859 insertions(+), 79 deletions(-) create mode 100644 sound/soc/codecs/ads117x.c create mode 100644 sound/soc/codecs/ads117x.h create mode 100644 sound/soc/omap/igep0020.c

2 1

[alsa-devel] [PATCH] ASoC: Add power down of widget powered up for suspend
by Joonyoung Shim 23 Nov '09

23 Nov '09

If the widget was powered up when the device enters to the suspend, it remains on because the power of the widget doesn't change, but we need to power down the widget on suspend. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/soc-dapm.c | 5 ++++- 1 files changed, 4 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d2af872..099d3b6 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -978,8 +978,11 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) continue; power = w->power_check(w); - if (power) + if (power) { sys_power = 1; + if (event == SND_SOC_DAPM_STREAM_SUSPEND) + power = 0; + } if (w->power == power) continue; -- 1.6.3.3

2 2

[alsa-devel] [PATCH 6/8] AI TB: Audio driver of Always Innovating OMAP3-based Touch Book
by Gregoire Gentil 23 Nov '09

23 Nov '09

Signed-off-by: Gregoire Gentil <gregoire(a)gentil.com> --- sound/soc/omap/Kconfig | 8 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/omap3touchbook.c | 211 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 221 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/omap3touchbook.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 653a362..c590252 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -83,6 +83,14 @@ config SND_OMAP_SOC_OMAP3_PANDORA help Say Y if you want to add support for SoC audio on the OMAP3 Pandora. +config SND_OMAP_SOC_OMAP3_TOUCHBOOK + tristate "SoC Audio support for OMAP3 Touch Book" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_TOUCHBOOK + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the OMAP3 Touch Book. + config SND_OMAP_SOC_OMAP3_BEAGLE tristate "SoC Audio support for OMAP3 Beagle" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_BEAGLE diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 02d6947..f360b82 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -15,6 +15,7 @@ snd-soc-omap3evm-objs := omap3evm.o snd-soc-sdp3430-objs := sdp3430.o snd-soc-omap3pandora-objs := omap3pandora.o snd-soc-omap3beagle-objs := omap3beagle.o +snd-soc-omap3touchbook-objs := omap3touchbook.o snd-soc-zoom2-objs := zoom2.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o @@ -26,4 +27,5 @@ obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(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_OMAP3_TOUCHBOOK) += snd-soc-omap3touchbook.o obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o diff --git a/sound/soc/omap/omap3touchbook.c b/sound/soc/omap/omap3touchbook.c new file mode 100644 index 0000000..584f07d --- /dev/null +++ b/sound/soc/omap/omap3touchbook.c @@ -0,0 +1,211 @@ +/* + * omap3touchbook.c -- SoC audio for OMAP3 Touch Book + * + * Author: Gregoire Gentil <gregoire(a)gentil.com> + * + * Derived from sound/soc/omap/omap3beagle.c + * + * 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/irq.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <plat/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +#define TB_HEADPHONE_GPIO 56 +#define TB_HEADPHONE_IRQ OMAP_GPIO_IRQ(TB_HEADPHONE_GPIO) + +static int omap3touchbook_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int fmt; + int ret; + + switch (params_channels(params)) { + case 2: /* Stereo I2S mode */ + fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + break; + case 4: /* Four channel TDM mode */ + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM; + break; + default: + return -EINVAL; + } + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops omap3touchbook_ops = { + .hw_params = omap3touchbook_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link omap3touchbook_dai = { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], + .ops = &omap3touchbook_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_omap3touchbook = { + .name = "omap3touchbook", + .platform = &omap_soc_platform, + .dai_link = &omap3touchbook_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device omap3touchbook_snd_devdata = { + .card = &snd_soc_omap3touchbook, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *omap3touchbook_snd_device; + +static void jack_work_func(struct work_struct *wq) +{ + struct snd_soc_device *socdev = platform_get_drvdata( + omap3touchbook_snd_device); + struct snd_soc_codec *codec = socdev->card->codec; + + snd_soc_dapm_sync(codec); +} +DECLARE_WORK(jack_work, jack_work_func); + +static irqreturn_t touchbook_headphone_event(int irq, void *snd) +{ + int status = gpio_get_value(TB_HEADPHONE_GPIO); + struct snd_soc_device *socdev = platform_get_drvdata( + omap3touchbook_snd_device); + struct snd_soc_codec *codec = socdev->card->codec; + + if (status) { + snd_soc_dapm_disable_pin(codec, "HFL"); + snd_soc_dapm_disable_pin(codec, "HFR"); + } else { + snd_soc_dapm_enable_pin(codec, "HFL"); + snd_soc_dapm_enable_pin(codec, "HFR"); + } + + schedule_work(&jack_work); + return IRQ_HANDLED; +} + +static int __init omap3touchbook_soc_init(void) +{ + int ret; + + if (!machine_is_touchbook()) { + pr_debug("Not OMAP3 Touch Book!\n"); + return -ENODEV; + } + pr_info("OMAP3 Touch Book SoC init\n"); + + omap3touchbook_snd_device = platform_device_alloc("soc-audio", -1); + if (!omap3touchbook_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(omap3touchbook_snd_device, + &omap3touchbook_snd_devdata); + omap3touchbook_snd_devdata.dev = &omap3touchbook_snd_device->dev; + *(unsigned int *)omap3touchbook_dai.cpu_dai->private_data = 1; + + /* Touch Book -- headphone jack sensor */ + omap_set_gpio_debounce(TB_HEADPHONE_GPIO, 1); + omap_set_gpio_debounce_time(TB_HEADPHONE_GPIO, 0xff); + + ret = request_irq(TB_HEADPHONE_IRQ, touchbook_headphone_event, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING, + "touchbook_headphone", + omap3touchbook_snd_device); + if (ret < 0) + goto err1; + + ret = platform_device_add(omap3touchbook_snd_device); + if (ret) + goto err1; + + /* Detect headphone status */ + touchbook_headphone_event(0, omap3touchbook_snd_device); + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(omap3touchbook_snd_device); + + return ret; +} + +static void __exit omap3touchbook_soc_exit(void) +{ + platform_device_unregister(omap3touchbook_snd_device); +} + +module_init(omap3touchbook_soc_init); +module_exit(omap3touchbook_soc_exit); + +MODULE_AUTHOR("Gregoire Gentil <gregoire(a)gentil.com>"); +MODULE_DESCRIPTION("ALSA SoC OMAP3 Touch Book"); +MODULE_LICENSE("GPL"); -- 1.6.0.4

2 1

[alsa-devel] Xonar Essence ST (PCI) on 2.6.32-rc8-00011-ga8a8a66: no analog audio, SPDIF fine
by Tony Vroon 23 Nov '09

23 Nov '09

Good evening, After the success of the Xonar D2X in my system, I decided to upgrade to an Essence ST (of the PCI variety, no D6 daughterboard). Unfortunately it seems that it produces no audio for me over the headphone port (the reason I bought this card) or the front-panel headphone port. I currently lack the RCA cabling to hook up the analog "speakers" port, but I promise to try this as soon as practical. The SPDIF output is functional and can be switched on and off using the mixer controls. I'm wondering if the Essence ST driver was written based on an a ST+D6 engineering sample as mine is the ST (no D6) retail box. Is it possible that the hardware is different? lspci output follows, in the hope that it is of use: 01:06.0 Multimedia audio controller [0401]: C-Media Electronics Inc CMI8788 [Oxygen HD Audio] [13f6:8788] Subsystem: ASUSTeK Computer Inc. Virtuoso 100 (Xonar ST) [1043:835d] Flags: bus master, medium devsel, latency 64, IRQ 17 I/O ports at c800 [size=256] Capabilities: [c0] Power Management version 2 Kernel driver in use: AV200 Also, from the ALSA side: # cat /proc/asound/ST/cmi8788 CMI8788 00: 80 e7 1e 86 d0 ed eb cf 00 00 88 bf ff 7f ff 3f 10: d8 ce 98 7e 7c ef c4 c1 00 00 8a bf 37 11 4f 04 20: 60 f7 93 bf af 5a 00 00 d7 4e 00 00 00 00 00 00 30: 30 81 91 92 20 92 ba 74 00 00 00 00 00 00 00 00 40: 10 00 00 00 10 40 00 00 00 00 00 00 00 00 00 00 50: 72 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 60: 19 01 00 09 19 01 00 09 00 00 00 00 00 00 00 00 70: 02 20 00 01 24 08 00 00 00 00 00 00 00 00 00 00 80: 00 40 4d 43 00 00 00 00 00 00 00 00 00 00 00 00 90: 98 11 bf 00 c0 01 00 00 00 00 00 00 00 00 00 00 a0: 00 00 00 00 01 00 93 01 8f 01 00 00 00 00 00 00 b0: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 c0: 20 e0 00 00 e4 00 00 00 00 00 00 00 00 00 00 00 d0: 10 00 03 00 00 00 00 00 00 00 00 00 1f 9f 96 00 e0: 01 00 00 00 07 00 0e 00 00 02 05 00 00 b2 02 00 f0: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 AC97 00: 0000 0000 0000 0000 0000 8000 0000 8808 10: 8808 8808 8808 9f1f 8808 0000 8000 0000 20: 0000 0000 0000 030c 0bc2 3c30 0000 0000 30: 0000 0000 0000 8080 8080 2000 0000 0000 40: 0000 0000 0000 0000 0000 0000 0000 0000 50: 0000 0000 0000 0000 0000 0000 0004 0000 60: 8080 188f 8043 0004 0000 0000 0000 0000 70: 0300 0011 c000 3935 0000 0000 434d 4969 At any rate, thank you for the oxygen/virtuoso driver. Hopefully this can be solved. The card does seem to be driven fine as the PCM sinks receive their audio without complaint and any multimedia player I use starts output. Regards, Tony V.

2 2

[PATCH] Davinci sound soc driver patch for Neuros OSD2 board support.
by Andrey Porodko 23 Nov '09

23 Nov '09

Signed-off-by: Andrey Porodko <panda(a)chelcom.ru> --- sound/soc/davinci/davinci-evm.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 67414f6..ffdd30e 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -55,6 +55,9 @@ static int evm_hw_params(struct snd_pcm_substream *substream, else if (machine_is_davinci_evm()) sysclk = 12288000; + else if (machine_is_neuros_osd2()) + sysclk = 27000000; + else if (machine_is_davinci_da830_evm() || machine_is_davinci_da850_evm()) sysclk = 24576000; @@ -246,6 +249,9 @@ static int __init evm_init(void) if (machine_is_davinci_evm()) { evm_snd_dev_data = &evm_snd_devdata; index = 0; + } else if (machine_is_neuros_osd2()) { + evm_snd_dev_data = &evm_snd_devdata; + index = 0; } else if (machine_is_davinci_dm355_evm()) { evm_snd_dev_data = &evm_snd_devdata; index = 1; -- 1.5.6.5 --------------000804080502060002040509--

1 0

[PATCH] Davinci sound soc driver patch for Neuros OSD2 board support.
by Andrey Porodko 23 Nov '09

23 Nov '09

Signed-off-by: Andrey Porodko <panda(a)chelcom.ru> --- sound/soc/davinci/davinci-evm.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 67414f6..ffdd30e 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -55,6 +55,9 @@ static int evm_hw_params(struct snd_pcm_substream *substream, else if (machine_is_davinci_evm()) sysclk = 12288000; + else if (machine_is_neuros_osd2()) + sysclk = 27000000; + else if (machine_is_davinci_da830_evm() || machine_is_davinci_da850_evm()) sysclk = 24576000; @@ -246,6 +249,9 @@ static int __init evm_init(void) if (machine_is_davinci_evm()) { evm_snd_dev_data = &evm_snd_devdata; index = 0; + } else if (machine_is_neuros_osd2()) { + evm_snd_dev_data = &evm_snd_devdata; + index = 0; } else if (machine_is_davinci_dm355_evm()) { evm_snd_dev_data = &evm_snd_devdata; index = 1; -- 1.5.6.5 --------------050305050402010500000508--

1 0

[alsa-devel] [PATCH 1/2] opti-miro: make miro.h header available outside the alsa directory
by Krzysztof Helt 23 Nov '09

23 Nov '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Move the miro.h header to the include/sound directory. It can be used in the Miro PCM20 radio driver (v4l). Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This is a part of my effort to restore the PCM20's radio driver. include/sound/aci.h | 73 ++++++++++++++++++++++++++++++++++++++++++++++ sound/isa/opti9xx/miro.c | 2 +- sound/isa/opti9xx/miro.h | 73 ---------------------------------------------- 3 files changed, 74 insertions(+), 74 deletions(-) create mode 100644 include/sound/aci.h delete mode 100644 sound/isa/opti9xx/miro.h diff --git a/include/sound/aci.h b/include/sound/aci.h new file mode 100644 index 0000000..bb796d0 --- /dev/null +++ b/include/sound/aci.h @@ -0,0 +1,73 @@ +#ifndef _ACI_H_ +#define _ACI_H_ + +#define ACI_REG_COMMAND 0 /* write register offset */ +#define ACI_REG_STATUS 1 /* read register offset */ +#define ACI_REG_BUSY 2 /* busy register offset */ +#define ACI_REG_RDS 2 /* PCM20: RDS register offset */ +#define ACI_MINTIME 500 /* ACI time out limit */ + +#define ACI_SET_MUTE 0x0d +#define ACI_SET_POWERAMP 0x0f +#define ACI_SET_TUNERMUTE 0xa3 +#define ACI_SET_TUNERMONO 0xa4 +#define ACI_SET_IDE 0xd0 +#define ACI_SET_WSS 0xd1 +#define ACI_SET_SOLOMODE 0xd2 +#define ACI_SET_PREAMP 0x03 +#define ACI_GET_PREAMP 0x21 +#define ACI_WRITE_TUNE 0xa7 +#define ACI_READ_TUNERSTEREO 0xa8 +#define ACI_READ_TUNERSTATION 0xa9 +#define ACI_READ_VERSION 0xf1 +#define ACI_READ_IDCODE 0xf2 +#define ACI_INIT 0xff +#define ACI_STATUS 0xf0 +#define ACI_S_GENERAL 0x00 +#define ACI_ERROR_OP 0xdf + +/* ACI Mixer */ + +/* These are the values for the right channel GET registers. + Add an offset of 0x01 for the left channel register. + (left=right+0x01) */ + +#define ACI_GET_MASTER 0x03 +#define ACI_GET_MIC 0x05 +#define ACI_GET_LINE 0x07 +#define ACI_GET_CD 0x09 +#define ACI_GET_SYNTH 0x0b +#define ACI_GET_PCM 0x0d +#define ACI_GET_LINE1 0x10 /* Radio on PCM20 */ +#define ACI_GET_LINE2 0x12 + +#define ACI_GET_EQ1 0x22 /* from Bass ... */ +#define ACI_GET_EQ2 0x24 +#define ACI_GET_EQ3 0x26 +#define ACI_GET_EQ4 0x28 +#define ACI_GET_EQ5 0x2a +#define ACI_GET_EQ6 0x2c +#define ACI_GET_EQ7 0x2e /* ... to Treble */ + +/* And these are the values for the right channel SET registers. + For left channel access you have to add an offset of 0x08. + MASTER is an exception, which needs an offset of 0x01 */ + +#define ACI_SET_MASTER 0x00 +#define ACI_SET_MIC 0x30 +#define ACI_SET_LINE 0x31 +#define ACI_SET_CD 0x34 +#define ACI_SET_SYNTH 0x33 +#define ACI_SET_PCM 0x32 +#define ACI_SET_LINE1 0x35 /* Radio on PCM20 */ +#define ACI_SET_LINE2 0x36 + +#define ACI_SET_EQ1 0x40 /* from Bass ... */ +#define ACI_SET_EQ2 0x41 +#define ACI_SET_EQ3 0x42 +#define ACI_SET_EQ4 0x43 +#define ACI_SET_EQ5 0x44 +#define ACI_SET_EQ6 0x45 +#define ACI_SET_EQ7 0x46 /* ... to Treble */ + +#endif /* _ACI_H_ */ diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index db4a4fb..932a067 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -40,7 +40,7 @@ #define SNDRV_LEGACY_FIND_FREE_IRQ #define SNDRV_LEGACY_FIND_FREE_DMA #include <sound/initval.h> -#include "miro.h" +#include <sound/aci.h> MODULE_AUTHOR("Martin Langer <martin-langer(a)gmx.de>"); MODULE_LICENSE("GPL"); diff --git a/sound/isa/opti9xx/miro.h b/sound/isa/opti9xx/miro.h deleted file mode 100644 index 6e1385b..0000000 --- a/sound/isa/opti9xx/miro.h +++ /dev/null @@ -1,73 +0,0 @@ -#ifndef _MIRO_H_ -#define _MIRO_H_ - -#define ACI_REG_COMMAND 0 /* write register offset */ -#define ACI_REG_STATUS 1 /* read register offset */ -#define ACI_REG_BUSY 2 /* busy register offset */ -#define ACI_REG_RDS 2 /* PCM20: RDS register offset */ -#define ACI_MINTIME 500 /* ACI time out limit */ - -#define ACI_SET_MUTE 0x0d -#define ACI_SET_POWERAMP 0x0f -#define ACI_SET_TUNERMUTE 0xa3 -#define ACI_SET_TUNERMONO 0xa4 -#define ACI_SET_IDE 0xd0 -#define ACI_SET_WSS 0xd1 -#define ACI_SET_SOLOMODE 0xd2 -#define ACI_SET_PREAMP 0x03 -#define ACI_GET_PREAMP 0x21 -#define ACI_WRITE_TUNE 0xa7 -#define ACI_READ_TUNERSTEREO 0xa8 -#define ACI_READ_TUNERSTATION 0xa9 -#define ACI_READ_VERSION 0xf1 -#define ACI_READ_IDCODE 0xf2 -#define ACI_INIT 0xff -#define ACI_STATUS 0xf0 -#define ACI_S_GENERAL 0x00 -#define ACI_ERROR_OP 0xdf - -/* ACI Mixer */ - -/* These are the values for the right channel GET registers. - Add an offset of 0x01 for the left channel register. - (left=right+0x01) */ - -#define ACI_GET_MASTER 0x03 -#define ACI_GET_MIC 0x05 -#define ACI_GET_LINE 0x07 -#define ACI_GET_CD 0x09 -#define ACI_GET_SYNTH 0x0b -#define ACI_GET_PCM 0x0d -#define ACI_GET_LINE1 0x10 /* Radio on PCM20 */ -#define ACI_GET_LINE2 0x12 - -#define ACI_GET_EQ1 0x22 /* from Bass ... */ -#define ACI_GET_EQ2 0x24 -#define ACI_GET_EQ3 0x26 -#define ACI_GET_EQ4 0x28 -#define ACI_GET_EQ5 0x2a -#define ACI_GET_EQ6 0x2c -#define ACI_GET_EQ7 0x2e /* ... to Treble */ - -/* And these are the values for the right channel SET registers. - For left channel access you have to add an offset of 0x08. - MASTER is an exception, which needs an offset of 0x01 */ - -#define ACI_SET_MASTER 0x00 -#define ACI_SET_MIC 0x30 -#define ACI_SET_LINE 0x31 -#define ACI_SET_CD 0x34 -#define ACI_SET_SYNTH 0x33 -#define ACI_SET_PCM 0x32 -#define ACI_SET_LINE1 0x35 /* Radio on PCM20 */ -#define ACI_SET_LINE2 0x36 - -#define ACI_SET_EQ1 0x40 /* from Bass ... */ -#define ACI_SET_EQ2 0x41 -#define ACI_SET_EQ3 0x42 -#define ACI_SET_EQ4 0x43 -#define ACI_SET_EQ5 0x44 -#define ACI_SET_EQ6 0x45 -#define ACI_SET_EQ7 0x46 /* ... to Treble */ - -#endif /* _MIRO_H_ */ -- 1.6.4 ---------------------------------------------------------------------- Internetowi nie placa! Otworz Konto Direct. Kliknij >> http://link.interia.pl/f2448

2 1

[alsa-devel] [PATCH] AACI cleanup
by Russell King - ARM Linux 23 Nov '09

23 Nov '09

Below is a patch which adds me to MAINTAINERS for this driver, and fixes the buffer size calculation to use the size which ALSA is expecting. Signed-off-by: Russell King <rmk+kernel(a)arm.linux.org.uk> -- Please review carefully, since I am presently unable to run-time test changes to this driver due to incompatibilities system ABI. MAINTAINERS | 5 +++++ sound/arm/aaci.c | 7 ++----- 2 files changed, 7 insertions(+), 5 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index c824b4d..dffb0c5 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -512,6 +512,11 @@ W: http://www.arm.linux.org.uk/ S: Maintained F: arch/arm/ +ARM PRIMECELL AACI PL041 DRIVER +M: Russell King <linux(a)arm.linux.org.uk> +S: Maintained +F: sound/arm/aaci.* + ARM PRIMECELL MMCI PL180/1 DRIVER S: Orphan F: drivers/mmc/host/mmci.* diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 1f0f821..a03fe80 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -18,10 +18,7 @@ #include <linux/interrupt.h> #include <linux/err.h> #include <linux/amba/bus.h> - -#include <asm/io.h> -#include <asm/irq.h> -#include <asm/sizes.h> +#include <linux/io.h> #include <sound/core.h> #include <sound/initval.h> @@ -534,7 +531,7 @@ static int aaci_pcm_prepare(struct snd_pcm_substream *substream) struct aaci_runtime *aacirun = runtime->private_data; aacirun->start = (void *)runtime->dma_area; - aacirun->end = aacirun->start + runtime->dma_bytes; + aacirun->end = aacirun->start + snd_pcm_lib_buffer_bytes(substream); aacirun->ptr = aacirun->start; aacirun->period = aacirun->bytes = frames_to_bytes(runtime, runtime->period_size);

2 2

[alsa-devel] [PATCH 2/2] opti-miro: expose ACI mixer to outside drivers
by Krzysztof Helt 22 Nov '09

22 Nov '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The ACI mixer is used to control the radio FM module installed on the Miro PCM20 sound card. Expose ACI mixer outside the sound card driver. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- These two patches are enough to implement basic functionality of the v4l driver for the PCM20's radio tuner. include/sound/aci.h | 17 +++ sound/isa/opti9xx/miro.c | 250 +++++++++++++++++++++++++++------------------- 2 files changed, 162 insertions(+), 105 deletions(-) diff --git a/include/sound/aci.h b/include/sound/aci.h index bb796d0..ee639d3 100644 --- a/include/sound/aci.h +++ b/include/sound/aci.h @@ -70,4 +70,21 @@ #define ACI_SET_EQ6 0x45 #define ACI_SET_EQ7 0x46 /* ... to Treble */ +struct snd_miro_aci { + unsigned long aci_port; + int aci_vendor; + int aci_product; + int aci_version; + int aci_amp; + int aci_preamp; + int aci_solomode; + + struct mutex aci_mutex; +}; + +int snd_aci_cmd(struct snd_miro_aci *aci, int write1, int write2, int write3); + +struct snd_miro_aci *snd_aci_get_aci(void); + #endif /* _ACI_H_ */ + diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index 932a067..40b64cd 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -96,7 +96,6 @@ MODULE_PARM_DESC(ide, "enable ide port"); #define OPTi9XX_MC_REG(n) n - struct snd_miro { unsigned short hardware; unsigned char password; @@ -120,17 +119,11 @@ struct snd_miro { long mpu_port; int mpu_irq; - unsigned long aci_port; - int aci_vendor; - int aci_product; - int aci_version; - int aci_amp; - int aci_preamp; - int aci_solomode; - - struct mutex aci_mutex; + struct snd_miro_aci *aci; }; +static struct snd_miro_aci aci_device; + static char * snd_opti9xx_names[] = { "unkown", "82C928", "82C929", @@ -142,13 +135,14 @@ static char * snd_opti9xx_names[] = { * ACI control */ -static int aci_busy_wait(struct snd_miro * miro) +static int aci_busy_wait(struct snd_miro_aci *aci) { long timeout; unsigned char byte; - for (timeout = 1; timeout <= ACI_MINTIME+30; timeout++) { - if (((byte=inb(miro->aci_port + ACI_REG_BUSY)) & 1) == 0) { + for (timeout = 1; timeout <= ACI_MINTIME + 30; timeout++) { + byte = inb(aci->aci_port + ACI_REG_BUSY); + if ((byte & 1) == 0) { if (timeout >= ACI_MINTIME) snd_printd("aci ready in round %ld.\n", timeout-ACI_MINTIME); @@ -174,10 +168,10 @@ static int aci_busy_wait(struct snd_miro * miro) return -EBUSY; } -static inline int aci_write(struct snd_miro * miro, unsigned char byte) +static inline int aci_write(struct snd_miro_aci *aci, unsigned char byte) { - if (aci_busy_wait(miro) >= 0) { - outb(byte, miro->aci_port + ACI_REG_COMMAND); + if (aci_busy_wait(aci) >= 0) { + outb(byte, aci->aci_port + ACI_REG_COMMAND); return 0; } else { snd_printk(KERN_ERR "aci busy, aci_write(0x%x) stopped.\n", byte); @@ -185,12 +179,12 @@ static inline int aci_write(struct snd_miro * miro, unsigned char byte) } } -static inline int aci_read(struct snd_miro * miro) +static inline int aci_read(struct snd_miro_aci *aci) { unsigned char byte; - if (aci_busy_wait(miro) >= 0) { - byte=inb(miro->aci_port + ACI_REG_STATUS); + if (aci_busy_wait(aci) >= 0) { + byte = inb(aci->aci_port + ACI_REG_STATUS); return byte; } else { snd_printk(KERN_ERR "aci busy, aci_read() stopped.\n"); @@ -198,39 +192,49 @@ static inline int aci_read(struct snd_miro * miro) } } -static int aci_cmd(struct snd_miro * miro, int write1, int write2, int write3) +int snd_aci_cmd(struct snd_miro_aci *aci, int write1, int write2, int write3) { int write[] = {write1, write2, write3}; int value, i; - if (mutex_lock_interruptible(&miro->aci_mutex)) + if (mutex_lock_interruptible(&aci->aci_mutex)) return -EINTR; for (i=0; i<3; i++) { if (write[i]< 0 || write[i] > 255) break; else { - value = aci_write(miro, write[i]); + value = aci_write(aci, write[i]); if (value < 0) goto out; } } - value = aci_read(miro); + value = aci_read(aci); -out: mutex_unlock(&miro->aci_mutex); +out: mutex_unlock(&aci->aci_mutex); return value; } +EXPORT_SYMBOL(snd_aci_cmd); + +static int aci_getvalue(struct snd_miro_aci *aci, unsigned char index) +{ + return snd_aci_cmd(aci, ACI_STATUS, index, -1); +} -static int aci_getvalue(struct snd_miro * miro, unsigned char index) +static int aci_setvalue(struct snd_miro_aci *aci, unsigned char index, + int value) { - return aci_cmd(miro, ACI_STATUS, index, -1); + return snd_aci_cmd(aci, index, value, -1); } -static int aci_setvalue(struct snd_miro * miro, unsigned char index, int value) +struct snd_miro_aci *snd_aci_get_aci(void) { - return aci_cmd(miro, index, value, -1); + if (aci_device.aci_port == 0) + return NULL; + return &aci_device; } +EXPORT_SYMBOL(snd_aci_get_aci); /* * MIXER part @@ -244,8 +248,10 @@ static int snd_miro_get_capture(struct snd_kcontrol *kcontrol, struct snd_miro *miro = snd_kcontrol_chip(kcontrol); int value; - if ((value = aci_getvalue(miro, ACI_S_GENERAL)) < 0) { - snd_printk(KERN_ERR "snd_miro_get_capture() failed: %d\n", value); + value = aci_getvalue(miro->aci, ACI_S_GENERAL); + if (value < 0) { + snd_printk(KERN_ERR "snd_miro_get_capture() failed: %d\n", + value); return value; } @@ -262,13 +268,15 @@ static int snd_miro_put_capture(struct snd_kcontrol *kcontrol, value = !(ucontrol->value.integer.value[0]); - if ((error = aci_setvalue(miro, ACI_SET_SOLOMODE, value)) < 0) { - snd_printk(KERN_ERR "snd_miro_put_capture() failed: %d\n", error); + error = aci_setvalue(miro->aci, ACI_SET_SOLOMODE, value); + if (error < 0) { + snd_printk(KERN_ERR "snd_miro_put_capture() failed: %d\n", + error); return error; } - change = (value != miro->aci_solomode); - miro->aci_solomode = value; + change = (value != miro->aci->aci_solomode); + miro->aci->aci_solomode = value; return change; } @@ -290,7 +298,7 @@ static int snd_miro_get_preamp(struct snd_kcontrol *kcontrol, struct snd_miro *miro = snd_kcontrol_chip(kcontrol); int value; - if (miro->aci_version <= 176) { + if (miro->aci->aci_version <= 176) { /* OSS says it's not readable with versions < 176. @@ -298,12 +306,14 @@ static int snd_miro_get_preamp(struct snd_kcontrol *kcontrol, which is a PCM12 with aci_version = 176. */ - ucontrol->value.integer.value[0] = miro->aci_preamp; + ucontrol->value.integer.value[0] = miro->aci->aci_preamp; return 0; } - if ((value = aci_getvalue(miro, ACI_GET_PREAMP)) < 0) { - snd_printk(KERN_ERR "snd_miro_get_preamp() failed: %d\n", value); + value = aci_getvalue(miro->aci, ACI_GET_PREAMP); + if (value < 0) { + snd_printk(KERN_ERR "snd_miro_get_preamp() failed: %d\n", + value); return value; } @@ -320,13 +330,15 @@ static int snd_miro_put_preamp(struct snd_kcontrol *kcontrol, value = ucontrol->value.integer.value[0]; - if ((error = aci_setvalue(miro, ACI_SET_PREAMP, value)) < 0) { - snd_printk(KERN_ERR "snd_miro_put_preamp() failed: %d\n", error); + error = aci_setvalue(miro->aci, ACI_SET_PREAMP, value); + if (error < 0) { + snd_printk(KERN_ERR "snd_miro_put_preamp() failed: %d\n", + error); return error; } - change = (value != miro->aci_preamp); - miro->aci_preamp = value; + change = (value != miro->aci->aci_preamp); + miro->aci->aci_preamp = value; return change; } @@ -337,7 +349,7 @@ static int snd_miro_get_amp(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_miro *miro = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = miro->aci_amp; + ucontrol->value.integer.value[0] = miro->aci->aci_amp; return 0; } @@ -350,13 +362,14 @@ static int snd_miro_put_amp(struct snd_kcontrol *kcontrol, value = ucontrol->value.integer.value[0]; - if ((error = aci_setvalue(miro, ACI_SET_POWERAMP, value)) < 0) { + error = aci_setvalue(miro->aci, ACI_SET_POWERAMP, value); + if (error < 0) { snd_printk(KERN_ERR "snd_miro_put_amp() to %d failed: %d\n", value, error); return error; } - change = (value != miro->aci_amp); - miro->aci_amp = value; + change = (value != miro->aci->aci_amp); + miro->aci->aci_amp = value; return change; } @@ -405,12 +418,14 @@ static int snd_miro_get_double(struct snd_kcontrol *kcontrol, int right_reg = kcontrol->private_value & 0xff; int left_reg = right_reg + 1; - if ((right_val = aci_getvalue(miro, right_reg)) < 0) { + right_val = aci_getvalue(miro->aci, right_reg); + if (right_val < 0) { snd_printk(KERN_ERR "aci_getvalue(%d) failed: %d\n", right_reg, right_val); return right_val; } - if ((left_val = aci_getvalue(miro, left_reg)) < 0) { + left_val = aci_getvalue(miro->aci, left_reg); + if (left_val < 0) { snd_printk(KERN_ERR "aci_getvalue(%d) failed: %d\n", left_reg, left_val); return left_val; } @@ -446,6 +461,7 @@ static int snd_miro_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_miro *miro = snd_kcontrol_chip(kcontrol); + struct snd_miro_aci *aci = miro->aci; int left, right, left_old, right_old; int setreg_left, setreg_right, getreg_left, getreg_right; int change, error; @@ -461,12 +477,14 @@ static int snd_miro_put_double(struct snd_kcontrol *kcontrol, getreg_right = kcontrol->private_value & 0xff; getreg_left = getreg_right + 1; - if ((left_old = aci_getvalue(miro, getreg_left)) < 0) { + left_old = aci_getvalue(aci, getreg_left); + if (left_old < 0) { snd_printk(KERN_ERR "aci_getvalue(%d) failed: %d\n", getreg_left, left_old); return left_old; } - if ((right_old = aci_getvalue(miro, getreg_right)) < 0) { + right_old = aci_getvalue(aci, getreg_right); + if (right_old < 0) { snd_printk(KERN_ERR "aci_getvalue(%d) failed: %d\n", getreg_right, right_old); return right_old; } @@ -485,13 +503,15 @@ static int snd_miro_put_double(struct snd_kcontrol *kcontrol, right_old = 0x80 - right_old; if (left >= 0) { - if ((error = aci_setvalue(miro, setreg_left, left)) < 0) { + error = aci_setvalue(aci, setreg_left, left); + if (error < 0) { snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", left, error); return error; } } else { - if ((error = aci_setvalue(miro, setreg_left, 0x80 - left)) < 0) { + error = aci_setvalue(aci, setreg_left, 0x80 - left); + if (error < 0) { snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", 0x80 - left, error); return error; @@ -499,13 +519,15 @@ static int snd_miro_put_double(struct snd_kcontrol *kcontrol, } if (right >= 0) { - if ((error = aci_setvalue(miro, setreg_right, right)) < 0) { + error = aci_setvalue(aci, setreg_right, right); + if (error < 0) { snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", right, error); return error; } } else { - if ((error = aci_setvalue(miro, setreg_right, 0x80 - right)) < 0) { + error = aci_setvalue(aci, setreg_right, 0x80 - right); + if (error < 0) { snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", 0x80 - right, error); return error; @@ -523,12 +545,14 @@ static int snd_miro_put_double(struct snd_kcontrol *kcontrol, left_old = 0x20 - left_old; right_old = 0x20 - right_old; - if ((error = aci_setvalue(miro, setreg_left, 0x20 - left)) < 0) { + error = aci_setvalue(aci, setreg_left, 0x20 - left); + if (error < 0) { snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", 0x20 - left, error); return error; } - if ((error = aci_setvalue(miro, setreg_right, 0x20 - right)) < 0) { + error = aci_setvalue(aci, setreg_right, 0x20 - right); + if (error < 0) { snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", 0x20 - right, error); return error; @@ -626,11 +650,13 @@ static unsigned char aci_init_values[][2] __devinitdata = { static int __devinit snd_set_aci_init_values(struct snd_miro *miro) { int idx, error; + struct snd_miro_aci *aci = miro->aci; /* enable WSS on PCM1 */ - if ((miro->aci_product == 'A') && wss) { - if ((error = aci_setvalue(miro, ACI_SET_WSS, wss)) < 0) { + if ((aci->aci_product == 'A') && wss) { + error = aci_setvalue(aci, ACI_SET_WSS, wss); + if (error < 0) { snd_printk(KERN_ERR "enabling WSS mode failed\n"); return error; } @@ -639,7 +665,8 @@ static int __devinit snd_set_aci_init_values(struct snd_miro *miro) /* enable IDE port */ if (ide) { - if ((error = aci_setvalue(miro, ACI_SET_IDE, ide)) < 0) { + error = aci_setvalue(aci, ACI_SET_IDE, ide); + if (error < 0) { snd_printk(KERN_ERR "enabling IDE port failed\n"); return error; } @@ -647,17 +674,18 @@ static int __devinit snd_set_aci_init_values(struct snd_miro *miro) /* set common aci values */ - for (idx = 0; idx < ARRAY_SIZE(aci_init_values); idx++) - if ((error = aci_setvalue(miro, aci_init_values[idx][0], - aci_init_values[idx][1])) < 0) { + for (idx = 0; idx < ARRAY_SIZE(aci_init_values); idx++) { + error = aci_setvalue(aci, aci_init_values[idx][0], + aci_init_values[idx][1]); + if (error < 0) { snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", aci_init_values[idx][0], error); return error; } - - miro->aci_amp = 0; - miro->aci_preamp = 0; - miro->aci_solomode = 1; + } + aci->aci_amp = 0; + aci->aci_preamp = 0; + aci->aci_solomode = 1; return 0; } @@ -688,7 +716,8 @@ static int __devinit snd_miro_mixer(struct snd_card *card, return err; } - if ((miro->aci_product == 'A') || (miro->aci_product == 'B')) { + if ((miro->aci->aci_product == 'A') || + (miro->aci->aci_product == 'B')) { /* PCM1/PCM12 with power-amp and Line 2 */ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_miro_line_control[0], miro))) < 0) return err; @@ -696,16 +725,17 @@ static int __devinit snd_miro_mixer(struct snd_card *card, return err; } - if ((miro->aci_product == 'B') || (miro->aci_product == 'C')) { + if ((miro->aci->aci_product == 'B') || + (miro->aci->aci_product == 'C')) { /* PCM12/PCM20 with mic-preamp */ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_miro_preamp_control[0], miro))) < 0) return err; - if (miro->aci_version >= 176) + if (miro->aci->aci_version >= 176) if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_miro_capture_control[0], miro))) < 0) return err; } - if (miro->aci_product == 'C') { + if (miro->aci->aci_product == 'C') { /* PCM20 with radio and 7 band equalizer */ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_miro_radio_control[0], miro))) < 0) return err; @@ -843,14 +873,15 @@ static void snd_miro_proc_read(struct snd_info_entry * entry, struct snd_info_buffer *buffer) { struct snd_miro *miro = (struct snd_miro *) entry->private_data; + struct snd_miro_aci *aci = miro->aci; char* model = "unknown"; /* miroSOUND PCM1 pro, early PCM12 */ if ((miro->hardware == OPTi9XX_HW_82C929) && - (miro->aci_vendor == 'm') && - (miro->aci_product == 'A')) { - switch(miro->aci_version) { + (aci->aci_vendor == 'm') && + (aci->aci_product == 'A')) { + switch (aci->aci_version) { case 3: model = "miroSOUND PCM1 pro"; break; @@ -863,9 +894,9 @@ static void snd_miro_proc_read(struct snd_info_entry * entry, /* miroSOUND PCM12, PCM12 (Rev. E), PCM12 pnp */ if ((miro->hardware == OPTi9XX_HW_82C924) && - (miro->aci_vendor == 'm') && - (miro->aci_product == 'B')) { - switch(miro->aci_version) { + (aci->aci_vendor == 'm') && + (aci->aci_product == 'B')) { + switch (aci->aci_version) { case 4: model = "miroSOUND PCM12"; break; @@ -881,9 +912,9 @@ static void snd_miro_proc_read(struct snd_info_entry * entry, /* miroSOUND PCM20 radio */ if ((miro->hardware == OPTi9XX_HW_82C924) && - (miro->aci_vendor == 'm') && - (miro->aci_product == 'C')) { - switch(miro->aci_version) { + (aci->aci_vendor == 'm') && + (aci->aci_product == 'C')) { + switch (aci->aci_version) { case 7: model = "miroSOUND PCM20 radio (Rev. E)"; break; @@ -907,17 +938,17 @@ static void snd_miro_proc_read(struct snd_info_entry * entry, snd_iprintf(buffer, "ACI information:\n"); snd_iprintf(buffer, " vendor : "); - switch(miro->aci_vendor) { + switch (aci->aci_vendor) { case 'm': snd_iprintf(buffer, "Miro\n"); break; default: - snd_iprintf(buffer, "unknown (0x%x)\n", miro->aci_vendor); + snd_iprintf(buffer, "unknown (0x%x)\n", aci->aci_vendor); break; } snd_iprintf(buffer, " product : "); - switch(miro->aci_product) { + switch (aci->aci_product) { case 'A': snd_iprintf(buffer, "miroSOUND PCM1 pro / (early) PCM12\n"); break; @@ -928,19 +959,19 @@ static void snd_miro_proc_read(struct snd_info_entry * entry, snd_iprintf(buffer, "miroSOUND PCM20 radio\n"); break; default: - snd_iprintf(buffer, "unknown (0x%x)\n", miro->aci_product); + snd_iprintf(buffer, "unknown (0x%x)\n", aci->aci_product); break; } snd_iprintf(buffer, " firmware: %d (0x%x)\n", - miro->aci_version, miro->aci_version); + aci->aci_version, aci->aci_version); snd_iprintf(buffer, " port : 0x%lx-0x%lx\n", - miro->aci_port, miro->aci_port+2); + aci->aci_port, aci->aci_port+2); snd_iprintf(buffer, " wss : 0x%x\n", wss); snd_iprintf(buffer, " ide : 0x%x\n", ide); - snd_iprintf(buffer, " solomode: 0x%x\n", miro->aci_solomode); - snd_iprintf(buffer, " amp : 0x%x\n", miro->aci_amp); - snd_iprintf(buffer, " preamp : 0x%x\n", miro->aci_preamp); + snd_iprintf(buffer, " solomode: 0x%x\n", aci->aci_solomode); + snd_iprintf(buffer, " amp : 0x%x\n", aci->aci_amp); + snd_iprintf(buffer, " preamp : 0x%x\n", aci->aci_preamp); } static void __devinit snd_miro_proc_init(struct snd_card *card, @@ -1139,46 +1170,53 @@ static int __devinit snd_card_miro_detect(struct snd_card *card, } static int __devinit snd_card_miro_aci_detect(struct snd_card *card, - struct snd_miro * miro) + struct snd_miro *miro) { unsigned char regval; int i; + struct snd_miro_aci *aci = &aci_device; + + miro->aci = aci; - mutex_init(&miro->aci_mutex); + mutex_init(&aci->aci_mutex); /* get ACI port from OPTi9xx MC 4 */ regval=inb(miro->mc_base + 4); - miro->aci_port = (regval & 0x10) ? 0x344: 0x354; + aci->aci_port = (regval & 0x10) ? 0x344 : 0x354; - if ((miro->res_aci_port = request_region(miro->aci_port, 3, "miro aci")) == NULL) { + miro->res_aci_port = request_region(aci->aci_port, 3, "miro aci"); + if (miro->res_aci_port == NULL) { snd_printk(KERN_ERR "aci i/o area 0x%lx-0x%lx already used.\n", - miro->aci_port, miro->aci_port+2); + aci->aci_port, aci->aci_port+2); return -ENOMEM; } /* force ACI into a known state */ for (i = 0; i < 3; i++) - if (aci_cmd(miro, ACI_ERROR_OP, -1, -1) < 0) { + if (snd_aci_cmd(aci, ACI_ERROR_OP, -1, -1) < 0) { snd_printk(KERN_ERR "can't force aci into known state.\n"); return -ENXIO; } - if ((miro->aci_vendor=aci_cmd(miro, ACI_READ_IDCODE, -1, -1)) < 0 || - (miro->aci_product=aci_cmd(miro, ACI_READ_IDCODE, -1, -1)) < 0) { - snd_printk(KERN_ERR "can't read aci id on 0x%lx.\n", miro->aci_port); + aci->aci_vendor = snd_aci_cmd(aci, ACI_READ_IDCODE, -1, -1); + aci->aci_product = snd_aci_cmd(aci, ACI_READ_IDCODE, -1, -1); + if (aci->aci_vendor < 0 || aci->aci_product < 0) { + snd_printk(KERN_ERR "can't read aci id on 0x%lx.\n", + aci->aci_port); return -ENXIO; } - if ((miro->aci_version=aci_cmd(miro, ACI_READ_VERSION, -1, -1)) < 0) { + aci->aci_version = snd_aci_cmd(aci, ACI_READ_VERSION, -1, -1); + if (aci->aci_version < 0) { snd_printk(KERN_ERR "can't read aci version on 0x%lx.\n", - miro->aci_port); + aci->aci_port); return -ENXIO; } - if (aci_cmd(miro, ACI_INIT, -1, -1) < 0 || - aci_cmd(miro, ACI_ERROR_OP, ACI_ERROR_OP, ACI_ERROR_OP) < 0 || - aci_cmd(miro, ACI_ERROR_OP, ACI_ERROR_OP, ACI_ERROR_OP) < 0) { + if (snd_aci_cmd(aci, ACI_INIT, -1, -1) < 0 || + snd_aci_cmd(aci, ACI_ERROR_OP, ACI_ERROR_OP, ACI_ERROR_OP) < 0 || + snd_aci_cmd(aci, ACI_ERROR_OP, ACI_ERROR_OP, ACI_ERROR_OP) < 0) { snd_printk(KERN_ERR "can't initialize aci.\n"); return -ENXIO; } @@ -1191,6 +1229,7 @@ static void snd_card_miro_free(struct snd_card *card) struct snd_miro *miro = card->private_data; release_and_free_resource(miro->res_aci_port); + miro->aci->aci_port = 0; release_and_free_resource(miro->res_mc_base); } @@ -1250,7 +1289,6 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) } miro->wss_base = port; - miro->mpu_port = mpu_port; miro->irq = irq; miro->mpu_irq = mpu_irq; miro->dma1 = dma1; @@ -1272,6 +1310,8 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) return -EBUSY; } } + miro->mpu_port = mpu_port; + if (miro->irq == SNDRV_AUTO_IRQ) { if ((miro->irq = snd_legacy_find_free_irq(possible_irqs)) < 0) { snd_card_free(card); @@ -1339,9 +1379,9 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) return error; } - if (miro->aci_vendor == 'm') { + if (miro->aci->aci_vendor == 'm') { /* It looks like a miro sound card. */ - switch (miro->aci_product) { + switch (miro->aci->aci_product) { case 'A': sprintf(card->shortname, "miroSOUND PCM1 pro / PCM12"); -- 1.6.4 ---------------------------------------------------------------------- Najlepsze oferty Sylwestrowe! Sprawdz >>> http://link.interia.pl/f2426

1 0

[alsa-devel] [PATCH] opti-miro: remove snd_card pointer from snd_miro structure
by Krzysztof Helt 21 Nov '09

21 Nov '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Remove the snd_card pointer from the snd_miro structure and do some small code improvements. Also, move Opti chipset detection before detection of the ACI mixer, so the mci_base value is set in one place only. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/opti9xx/miro.c | 55 ++++++++++++++++++++-------------------------- 1 files changed, 24 insertions(+), 31 deletions(-) diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index b8170ad..db4a4fb 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -110,7 +110,6 @@ struct snd_miro { unsigned long pwd_reg; spinlock_t lock; - struct snd_card *card; struct snd_pcm *pcm; long wss_base; @@ -132,8 +131,6 @@ struct snd_miro { struct mutex aci_mutex; }; -static void snd_miro_proc_init(struct snd_miro * miro); - static char * snd_opti9xx_names[] = { "unkown", "82C928", "82C929", @@ -457,11 +454,9 @@ static int snd_miro_put_double(struct snd_kcontrol *kcontrol, right = ucontrol->value.integer.value[1]; setreg_right = (kcontrol->private_value >> 8) & 0xff; - if (setreg_right == ACI_SET_MASTER) { - setreg_left = setreg_right + 1; - } else { - setreg_left = setreg_right + 8; - } + setreg_left = setreg_right + 8; + if (setreg_right == ACI_SET_MASTER) + setreg_left -= 7; getreg_right = kcontrol->private_value & 0xff; getreg_left = getreg_right + 1; @@ -667,17 +662,15 @@ static int __devinit snd_set_aci_init_values(struct snd_miro *miro) return 0; } -static int __devinit snd_miro_mixer(struct snd_miro *miro) +static int __devinit snd_miro_mixer(struct snd_card *card, + struct snd_miro *miro) { - struct snd_card *card; unsigned int idx; int err; - if (snd_BUG_ON(!miro || !miro->card)) + if (snd_BUG_ON(!miro || !card)) return -EINVAL; - card = miro->card; - switch (miro->hardware) { case OPTi9XX_HW_82C924: strcpy(card->mixername, "ACI & OPTi924"); @@ -950,11 +943,12 @@ static void snd_miro_proc_read(struct snd_info_entry * entry, snd_iprintf(buffer, " preamp : 0x%x\n", miro->aci_preamp); } -static void __devinit snd_miro_proc_init(struct snd_miro * miro) +static void __devinit snd_miro_proc_init(struct snd_card *card, + struct snd_miro *miro) { struct snd_info_entry *entry; - if (! snd_card_proc_new(miro->card, "miro", &entry)) + if (!snd_card_proc_new(card, "miro", &entry)) snd_info_set_text_ops(entry, miro, snd_miro_proc_read); } @@ -971,20 +965,18 @@ static int __devinit snd_miro_configure(struct snd_miro *chip) unsigned char mpu_irq_bits; unsigned long flags; + snd_miro_write_mask(chip, OPTi9XX_MC_REG(1), 0x80, 0x80); + snd_miro_write_mask(chip, OPTi9XX_MC_REG(2), 0x20, 0x20); /* OPL4 */ + snd_miro_write_mask(chip, OPTi9XX_MC_REG(5), 0x02, 0x02); + switch (chip->hardware) { case OPTi9XX_HW_82C924: snd_miro_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x02); - snd_miro_write_mask(chip, OPTi9XX_MC_REG(1), 0x80, 0x80); - snd_miro_write_mask(chip, OPTi9XX_MC_REG(2), 0x20, 0x20); /* OPL4 */ snd_miro_write_mask(chip, OPTi9XX_MC_REG(3), 0xf0, 0xff); - snd_miro_write_mask(chip, OPTi9XX_MC_REG(5), 0x02, 0x02); break; case OPTi9XX_HW_82C929: /* untested init commands for OPTi929 */ - snd_miro_write_mask(chip, OPTi9XX_MC_REG(1), 0x80, 0x80); - snd_miro_write_mask(chip, OPTi9XX_MC_REG(2), 0x20, 0x20); /* OPL4 */ snd_miro_write_mask(chip, OPTi9XX_MC_REG(4), 0x00, 0x0c); - snd_miro_write_mask(chip, OPTi9XX_MC_REG(5), 0x02, 0x02); break; default: snd_printk(KERN_ERR "chip %d not supported\n", chip->hardware); @@ -1156,7 +1148,6 @@ static int __devinit snd_card_miro_aci_detect(struct snd_card *card, /* get ACI port from OPTi9xx MC 4 */ - miro->mc_base = 0xf8c; regval=inb(miro->mc_base + 4); miro->aci_port = (regval & 0x10) ? 0x344: 0x354; @@ -1232,7 +1223,13 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) card->private_free = snd_card_miro_free; miro = card->private_data; - miro->card = card; + + error = snd_card_miro_detect(card, miro); + if (error < 0) { + snd_card_free(card); + snd_printk(KERN_ERR "unable to detect OPTi9xx chip\n"); + return -ENODEV; + } if ((error = snd_card_miro_aci_detect(card, miro)) < 0) { snd_card_free(card); @@ -1241,13 +1238,8 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) } /* init proc interface */ - snd_miro_proc_init(miro); + snd_miro_proc_init(card, miro); - if ((error = snd_card_miro_detect(card, miro)) < 0) { - snd_card_free(card); - snd_printk(KERN_ERR "unable to detect OPTi9xx chip\n"); - return -ENODEV; - } if (! miro->res_mc_base && (miro->res_mc_base = request_region(miro->mc_base, miro->mc_base_size, @@ -1341,7 +1333,8 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) miro->pcm = pcm; - if ((error = snd_miro_mixer(miro)) < 0) { + error = snd_miro_mixer(card, miro); + if (error < 0) { snd_card_free(card); return error; } -- 1.6.4 ---------------------------------------------------------------------- Wygraj pobyt w Alpach dla ca�ej rodziny Kliknij >>> http://link.interia.pl/f2446

2 1

[alsa-devel] McBSP troubles
by Sergey Lapin 21 Nov '09

21 Nov '09

Hi, all! I have problems making any sound with current linux-omap master. ASoC repository did not change anything either. I use slightly modified (removed all GPIOs/jacks and different clock names because we use OMAP3 instead of OMAP2 n810 uses) sound/soc/omap/n810.c, with McBSP2. Codec is tlv320aic32 and it is well supplied with 13 MHz clock, so should work happily. Old 2.6.28-omap1 kernel + modifications works well with this, but it lacks some fundamental features recently added to mainline and linux-omap kernels, so we considered upgrading. Almost all stuff on custom board works, except for sound, which I fight for several weeks already, so I look for any help with this. I use OMAP3525 SoC. Debugging output of mcbsp driver is the following: gnome5:~# aplay -c 2 -f S16_LE /dev/urandom [ 4593.488159] disabling LINE_OUT [ 4593.491302] disabling DMIC [ 4593.494018] Disabling ASSYM pin Playing raw data '/dev/urandom' : Signed 16 bit Little Endian, Rate 8000 Hz, Stereo [ 4593.620086] omap-mcbsp omap-mcbsp.2: Configuring McBSP2 phys_base: 0x49022000 [ 4593.678375] gnome5_jack_event [ 4593.735015] omap-mcbsp omap-mcbsp.2: **** McBSP2 regs **** [ 4593.740539] omap-mcbsp omap-mcbsp.2: DRR2: 0xd907da5a [ 4593.745697] omap-mcbsp omap-mcbsp.2: DRR1: 0x0000 [ 4593.750518] omap-mcbsp omap-mcbsp.2: DXR2: 0x0000 [ 4593.755340] omap-mcbsp omap-mcbsp.2: DXR1: 0x0000 [ 4593.760162] omap-mcbsp omap-mcbsp.2: SPCR2: 0x02f5 [ 4593.764984] omap-mcbsp omap-mcbsp.2: SPCR1: 0x0030 [ 4593.769805] omap-mcbsp omap-mcbsp.2: RCR2: 0x8041 [ 4593.774627] omap-mcbsp omap-mcbsp.2: RCR1: 0x0040 [ 4593.779449] omap-mcbsp omap-mcbsp.2: XCR2: 0x8041 [ 4593.784271] omap-mcbsp omap-mcbsp.2: XCR1: 0x0040 [ 4593.789093] omap-mcbsp omap-mcbsp.2: SRGR2: 0x001f [ 4593.793884] omap-mcbsp omap-mcbsp.2: SRGR1: 0x0f00 [ 4593.798706] omap-mcbsp omap-mcbsp.2: PCR0: 0x000f [ 4593.803527] omap-mcbsp omap-mcbsp.2: *********************** [ 4603.811187] ALSA sound/core/pcm_lib.c:1708: playback write error (DMA or IRQ trouble?) [ 4614.818878] ALSA sound/core/pcm_lib.c:1708: playback write error (DMA or IRQ trouble?) aplay: pcm_write:1442: write error: And no sound at all. Codec is supplied properly with clock, and programmed normally, so I think that's McBSP trouble, but I might be wrong. Clock is supplied properly. So, what is changed and needs updating? Is it McBSP or codec? I see no signals using oscilloscope on _FSX and _CLKX. Any ideas? S.

1 1

[alsa-devel] Sigmatel STAC 9228
by Alonso Quiroz H. 21 Nov '09

21 Nov '09

Hey guys!, this is my first mail to the list. Excuse me if this is not the better way to send my request. I have a Dell XPS M1530 laptop, with a Sigmatel soundcard... well, read it for yourselves: http://pastebin.ca/1679794 By default (in Ubuntu 9.10) the card doesn't work in 5.1 mode, and that is my problem... by default it doesn't show the option "Surround 5.1" I have tried a lot of combinations of model=XXXXX in /etc/modprobe.d/alsa-base.conf, starting by the ones in /usr/share/doc/alsa-base/driver/HD-Audio-Models.txt. The better result is with "3stack": It 's possible to select the 5.1 option, but I don't get sound at all. The weird thing is that de volume control of Pulse shows bars moving. Thanks in advance. Sorry about my english. Alonso Quiroz H.

1 0

[alsa-devel] RFC: ice1712 virtual devices
by Arno Schuring 21 Nov '09

21 Nov '09

Hello list, This is basically a resend of http://thread.gmane.org/gmane.linux.alsa.devel/59481/focus=59672 , which fixed the front: device of ice1712 cards to accept two-channel input. Currently, the front: device is exposed through the route plugin, which requires all clients to mmap all 10 channels, even though the front device is supposed to be a stereo device. This patch changes the front: device definition such that it matches the definition of iec958 in the same file. Additionally, I'm tempted to remove the surround* definitions because the chip does not really offer surround-style multichannel: it basically just offers multiple stereo channels, and does not provide any channel mapping beyond stereo. Finally, I'm also experimenting with the dshare plugin to allow applications to access the iec958: and front: devices simultaneously. Can anyone point me to a working example for this? From reading the alsa-lib documentation, it is not clear to me how I should nest the different plugins. Many thanks, Arno Schuring -- diff --git a/src/conf/cards/ICE1712.conf b/src/conf/cards/ICE1712.conf index 01e50d2..d7acb81 100644 --- a/src/conf/cards/ICE1712.conf +++ b/src/conf/cards/ICE1712.conf @@ -32,12 +32,28 @@ ICE1712.pcm.front.0 { @args.CARD { type string } - type route - ttable.0.0 1 - ttable.1.1 1 - slave.pcm { - type hw - card $CARD + type asym + playback.pcm { + type route + ttable.0.0 1 + ttable.1.1 1 + slave.pcm { + type hw + card $CARD + } + slave.format S32_LE + slave.channels 10 + } + capture.pcm { + type route + ttable.0.0 1 + ttable.1.1 1 + slave.pcm { + type hw + card $CARD + } + slave.format S32_LE + slave.channels 12 } }

4 7

[alsa-devel] opti-miro: fix missing semicolon
by Krzysztof Helt 20 Nov '09

20 Nov '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Fix the mistake introduced by me earlier patch. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- I have borrowed the Miro PCM20 card. I can test the driver while making patches. diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index b8170ad..1776103 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -1277,7 +1277,7 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) if (mpu_port < 0) { snd_card_free(card); snd_printk(KERN_ERR "unable to find a free MPU401 port\n"); - return -EBUSY + return -EBUSY; } } if (miro->irq == SNDRV_AUTO_IRQ) { ---------------------------------------------------------------------- KONKURS! Kliknij i wygraj luksusowe auto na weekend lub jedn� z 650 innych nagr�d od L’Oreal Men Expert!! >> http://link.interia.pl/f2428

2 1

[alsa-devel] ALSA: hda - add possibility to choose speakers configuration for 4930g
by Łukasz Wojniłowicz 20 Nov '09

20 Nov '09

As in wrote title patch adds possibility to choose speakers configuration for 4930g. This feature was available in ALSA 1.0.19 (or even 1.0.20) but disappeared in 1.0.21

2 1

[alsa-devel] Testers wanted: New SB X-Fi driver
by Insaen 20 Nov '09

20 Nov '09

I own a SB x-fi fatality incl. the midi input box witch i realy would like to use for my bass guitar and use the record options, i run the card on ubuntu 9.04 and its running well but can't realy set the jack system right for software like rosegarden or lmms or any other music editing software. I know it's got something to do with my alsa config?? is there a forum about this toppick?Simon says have a nice day..2-;

1 0

[alsa-devel] ALSA: hda - 4930g switch lfe with side in mixer
by Łukasz Wojniłowicz 20 Nov '09

20 Nov '09

Built-in sub-woofer can now be controlled by lfe slider instead of side slider on Acer Aspire 5930g

1 0

[alsa-devel] [PATCH] sound: usbmidi: Use hweight16
by Akinobu Mita 20 Nov '09

20 Nov '09

Use hweight16 instead of Brian Kernighan's/Peter Wegner's method Signed-off-by: Akinobu Mita <akinobu.mita(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org Cc: Clemens Ladisch <clemens(a)ladisch.de> --- sound/usb/usbmidi.c | 13 ++----------- 1 files changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 0eff19c..e5b0689 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -1062,15 +1062,6 @@ static int snd_usbmidi_in_endpoint_create(struct snd_usb_midi* umidi, return 0; } -static unsigned int snd_usbmidi_count_bits(unsigned int x) -{ - unsigned int bits; - - for (bits = 0; x; ++bits) - x &= x - 1; - return bits; -} - /* * Frees an output endpoint. * May be called when ep hasn't been initialized completely. @@ -1914,8 +1905,8 @@ int snd_usb_create_midi_interface(struct snd_usb_audio* chip, out_ports = 0; in_ports = 0; for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) { - out_ports += snd_usbmidi_count_bits(endpoints[i].out_cables); - in_ports += snd_usbmidi_count_bits(endpoints[i].in_cables); + out_ports += hweight16(endpoints[i].out_cables); + in_ports += hweight16(endpoints[i].in_cables); } err = snd_usbmidi_create_rawmidi(umidi, out_ports, in_ports); if (err < 0) { -- 1.6.5.1

2 1

[alsa-devel] Mute lfe and side for 4930g when pluging in headphones
by Łukasz Wojniłowicz 20 Nov '09

20 Nov '09

Sorry that I'm not sending this patch by git but it takes for me too long to clone alsa-kernel.git repository at 15 KB download rate. Thanks for your understanding

2 1

[alsa-devel] [PATCH 1/4] ASoC: DaVinci: i2s, reduce underruns by combining into 1 element
by Troy Kisky 19 Nov '09

19 Nov '09

Allow the left and right 16 bit samples to be shifted out as 1 32 bit sample. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> --- This applies to Marks for-2.6.33 branch Sorry about the prior bad patch base. arch/arm/mach-davinci/include/mach/asp.h | 6 ++ sound/soc/davinci/davinci-i2s.c | 74 ++++++++++++++++++++++-------- 2 files changed, 61 insertions(+), 19 deletions(-) diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 18e4ce3..019c647 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -51,6 +51,12 @@ struct snd_platform_data { u32 rx_dma_offset; enum dma_event_q eventq_no; /* event queue number */ unsigned int codec_fmt; + /* + * Allowing this is more efficient and eliminates left and right swaps + * caused by underruns, but will swap the left and right channels + * when compared to previous behavior. + */ + unsigned enable_channel_combine:1; /* McASP specific fields */ int tdm_slots; diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index d336786..b2a5372 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -97,6 +97,23 @@ enum { DAVINCI_MCBSP_WORD_32, }; +static const unsigned char data_type[SNDRV_PCM_FORMAT_S32_LE + 1] = { + [SNDRV_PCM_FORMAT_S8] = 1, + [SNDRV_PCM_FORMAT_S16_LE] = 2, + [SNDRV_PCM_FORMAT_S32_LE] = 4, +}; + +static const unsigned char asp_word_length[SNDRV_PCM_FORMAT_S32_LE + 1] = { + [SNDRV_PCM_FORMAT_S8] = DAVINCI_MCBSP_WORD_8, + [SNDRV_PCM_FORMAT_S16_LE] = DAVINCI_MCBSP_WORD_16, + [SNDRV_PCM_FORMAT_S32_LE] = DAVINCI_MCBSP_WORD_32, +}; + +static const unsigned char double_fmt[SNDRV_PCM_FORMAT_S32_LE + 1] = { + [SNDRV_PCM_FORMAT_S8] = SNDRV_PCM_FORMAT_S16_LE, + [SNDRV_PCM_FORMAT_S16_LE] = SNDRV_PCM_FORMAT_S32_LE, +}; + struct davinci_mcbsp_dev { struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; @@ -105,6 +122,27 @@ struct davinci_mcbsp_dev { int mode; u32 pcr; struct clk *clk; + /* + * Combining both channels into 1 element will at least double the + * amount of time between servicing the dma channel, increase + * effiency, and reduce the chance of overrun/underrun. But, + * it will result in the left & right channels being swapped. + * + * If relabeling the left and right channels is not possible, + * you may want to let the codec know to swap them back. + * + * It may allow x10 the amount of time to service dma requests, + * if the codec is master and is using an unnecessarily fast bit clock + * (ie. tlvaic23b), independent of the sample rate. So, having an + * entire frame at once means it can be serviced at the sample rate + * instead of the bit clock rate. + * + * In the now unlikely case that an underrun still + * occurs, both the left and right samples will be repeated + * so that no pops are heard, and the left and right channels + * won't end up being swapped because of the underrun. + */ + unsigned enable_channel_combine:1; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -344,6 +382,8 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, int mcbsp_word_length; unsigned int rcr, xcr, srgr; u32 spcr; + snd_pcm_format_t fmt; + unsigned element_cnt = 1; /* general line settings */ spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); @@ -373,29 +413,24 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, xcr |= DAVINCI_MCBSP_XCR_XDATDLY(1); } /* Determine xfer data type */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - dma_params->data_type = 1; - mcbsp_word_length = DAVINCI_MCBSP_WORD_8; - break; - case SNDRV_PCM_FORMAT_S16_LE: - dma_params->data_type = 2; - mcbsp_word_length = DAVINCI_MCBSP_WORD_16; - break; - case SNDRV_PCM_FORMAT_S32_LE: - dma_params->data_type = 4; - mcbsp_word_length = DAVINCI_MCBSP_WORD_32; - break; - default: + fmt = params_format(params); + if ((fmt > SNDRV_PCM_FORMAT_S32_LE) || !data_type[fmt]) { printk(KERN_WARNING "davinci-i2s: unsupported PCM format\n"); return -EINVAL; } - dma_params->acnt = dma_params->data_type; + if (params_channels(params) == 2) { + element_cnt = 2; + if (double_fmt[fmt] && dev->enable_channel_combine) { + element_cnt = 1; + fmt = double_fmt[fmt]; + } + } + dma_params->acnt = dma_params->data_type = data_type[fmt]; dma_params->fifo_level = 0; - - rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(1); - xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(1); + mcbsp_word_length = asp_word_length[fmt]; + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); rcr |= DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length); @@ -510,7 +545,8 @@ static int davinci_i2s_probe(struct platform_device *pdev) ret = -ENOMEM; goto err_release_region; } - + if (pdata) + dev->enable_channel_combine = pdata->enable_channel_combine; dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { ret = -ENODEV; -- 1.5.6.3

2 5

[alsa-devel] Model change suggestion for Acer Aspire 5930g
by Claudio Viano 19 Nov '09

19 Nov '09

Hello, this message to suggest a setting change for the Acer Aspire 5930g laptop and its Intel HDA ALC888 card. Currently Alsa defaults to model=acer-aspire-4930g for this laptop, but with that option the tuba bass does not work, it works only upon initial installation of the OS and even then it does not get muted upon headphones plug-in. I tested that using model=acer-aspire-6530g instead everything works: the tuba bass gets muted along with the other built-in speakers upon headphones insertion, the internal mic works perfectly etc. I see in the change log for 1.0.21 that the Aspire 6930g has been migrated from model 4930g to 6530g. The 5930g should go the same way. There's not much more to it really, if you need any more information do not hesitate to ask me, I hope this is the right channel for this communication. I attach the output of alsa-info. Kind regards Claudio Viano

2 1

[alsa-devel] [PATCH] cs4236: add dB scale for all volume controls
by Krzysztof Helt 19 Nov '09

19 Nov '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Use db scale for all volume controls according to Crystal's datasheets. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/cs423x/cs4236_lib.c | 152 +++++++++++++++++++++++++++++------------ 1 files changed, 108 insertions(+), 44 deletions(-) diff --git a/sound/isa/cs423x/cs4236_lib.c b/sound/isa/cs423x/cs4236_lib.c index 4c4024a..c5adca3 100644 --- a/sound/isa/cs423x/cs4236_lib.c +++ b/sound/isa/cs423x/cs4236_lib.c @@ -88,6 +88,7 @@ #include <sound/wss.h> #include <sound/asoundef.h> #include <sound/initval.h> +#include <sound/tlv.h> /* * @@ -399,6 +400,14 @@ int snd_cs4236_pcm(struct snd_wss *chip, int device, struct snd_pcm **rpcm) .get = snd_cs4236_get_single, .put = snd_cs4236_put_single, \ .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) } +#define CS4236_SINGLE_TLV(xname, xindex, reg, shift, mask, invert, xtlv) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .info = snd_cs4236_info_single, \ + .get = snd_cs4236_get_single, .put = snd_cs4236_put_single, \ + .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24), \ + .tlv = { .p = (xtlv) } } + static int snd_cs4236_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { int mask = (kcontrol->private_value >> 16) & 0xff; @@ -502,6 +511,16 @@ static int snd_cs4236_put_singlec(struct snd_kcontrol *kcontrol, struct snd_ctl_ .get = snd_cs4236_get_double, .put = snd_cs4236_put_double, \ .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22) } +#define CS4236_DOUBLE_TLV(xname, xindex, left_reg, right_reg, shift_left, \ + shift_right, mask, invert, xtlv) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .info = snd_cs4236_info_double, \ + .get = snd_cs4236_get_double, .put = snd_cs4236_put_double, \ + .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | \ + (shift_right << 19) | (mask << 24) | (invert << 22), \ + .tlv = { .p = (xtlv) } } + static int snd_cs4236_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { int mask = (kcontrol->private_value >> 24) & 0xff; @@ -572,12 +591,23 @@ static int snd_cs4236_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_e return change; } -#define CS4236_DOUBLE1(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \ +#define CS4236_DOUBLE1(xname, xindex, left_reg, right_reg, shift_left, \ + shift_right, mask, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ .info = snd_cs4236_info_double, \ .get = snd_cs4236_get_double1, .put = snd_cs4236_put_double1, \ .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22) } +#define CS4236_DOUBLE1_TLV(xname, xindex, left_reg, right_reg, shift_left, \ + shift_right, mask, invert, xtlv) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .info = snd_cs4236_info_double, \ + .get = snd_cs4236_get_double1, .put = snd_cs4236_put_double1, \ + .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | \ + (shift_right << 19) | (mask << 24) | (invert << 22), \ + .tlv = { .p = (xtlv) } } + static int snd_cs4236_get_double1(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_wss *chip = snd_kcontrol_chip(kcontrol); @@ -631,16 +661,18 @@ static int snd_cs4236_put_double1(struct snd_kcontrol *kcontrol, struct snd_ctl_ return change; } -#define CS4236_MASTER_DIGITAL(xname, xindex) \ +#define CS4236_MASTER_DIGITAL(xname, xindex, xtlv) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ .info = snd_cs4236_info_double, \ .get = snd_cs4236_get_master_digital, .put = snd_cs4236_put_master_digital, \ - .private_value = 71 << 24 } + .private_value = 71 << 24, \ + .tlv = { .p = (xtlv) } } static inline int snd_cs4236_mixer_master_digital_invert_volume(int vol) { return (vol < 64) ? 63 - vol : 64 + (71 - vol); -} +} static int snd_cs4236_get_master_digital(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -673,11 +705,13 @@ static int snd_cs4236_put_master_digital(struct snd_kcontrol *kcontrol, struct s return change; } -#define CS4235_OUTPUT_ACCU(xname, xindex) \ +#define CS4235_OUTPUT_ACCU(xname, xindex, xtlv) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ .info = snd_cs4236_info_double, \ .get = snd_cs4235_get_output_accu, .put = snd_cs4235_put_output_accu, \ - .private_value = 3 << 24 } + .private_value = 3 << 24, \ + .tlv = { .p = (xtlv) } } static inline int snd_cs4235_mixer_output_accu_get_volume(int vol) { @@ -732,41 +766,56 @@ static int snd_cs4235_put_output_accu(struct snd_kcontrol *kcontrol, struct snd_ return change; } +static const DECLARE_TLV_DB_SCALE(db_scale_7bit, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_6bit, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_6bit_12db_max, -8250, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_5bit_12db_max, -3450, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_5bit_22db_max, -2400, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_4bit, -4500, 300, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_2bit, -1800, 600, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_rec_gain, 0, 150, 0); + static struct snd_kcontrol_new snd_cs4236_controls[] = { CS4236_DOUBLE("Master Digital Playback Switch", 0, CS4236_LEFT_MASTER, CS4236_RIGHT_MASTER, 7, 7, 1, 1), CS4236_DOUBLE("Master Digital Capture Switch", 0, CS4236_DAC_MUTE, CS4236_DAC_MUTE, 7, 6, 1, 1), -CS4236_MASTER_DIGITAL("Master Digital Volume", 0), +CS4236_MASTER_DIGITAL("Master Digital Volume", 0, db_scale_7bit), -CS4236_DOUBLE("Capture Boost Volume", 0, - CS4236_LEFT_MIX_CTRL, CS4236_RIGHT_MIX_CTRL, 5, 5, 3, 1), +CS4236_DOUBLE_TLV("Capture Boost Volume", 0, + CS4236_LEFT_MIX_CTRL, CS4236_RIGHT_MIX_CTRL, 5, 5, 3, 1, + db_scale_2bit), WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), -WSS_DOUBLE("PCM Playback Volume", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1), +WSS_DOUBLE_TLV("PCM Playback Volume", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1, + db_scale_6bit), CS4236_DOUBLE("DSP Playback Switch", 0, CS4236_LEFT_DSP, CS4236_RIGHT_DSP, 7, 7, 1, 1), -CS4236_DOUBLE("DSP Playback Volume", 0, - CS4236_LEFT_DSP, CS4236_RIGHT_DSP, 0, 0, 63, 1), +CS4236_DOUBLE_TLV("DSP Playback Volume", 0, + CS4236_LEFT_DSP, CS4236_RIGHT_DSP, 0, 0, 63, 1, + db_scale_6bit), CS4236_DOUBLE("FM Playback Switch", 0, CS4236_LEFT_FM, CS4236_RIGHT_FM, 7, 7, 1, 1), -CS4236_DOUBLE("FM Playback Volume", 0, - CS4236_LEFT_FM, CS4236_RIGHT_FM, 0, 0, 63, 1), +CS4236_DOUBLE_TLV("FM Playback Volume", 0, + CS4236_LEFT_FM, CS4236_RIGHT_FM, 0, 0, 63, 1, + db_scale_6bit), CS4236_DOUBLE("Wavetable Playback Switch", 0, CS4236_LEFT_WAVE, CS4236_RIGHT_WAVE, 7, 7, 1, 1), -CS4236_DOUBLE("Wavetable Playback Volume", 0, - CS4236_LEFT_WAVE, CS4236_RIGHT_WAVE, 0, 0, 63, 1), +CS4236_DOUBLE_TLV("Wavetable Playback Volume", 0, + CS4236_LEFT_WAVE, CS4236_RIGHT_WAVE, 0, 0, 63, 1, + db_scale_6bit_12db_max), WSS_DOUBLE("Synth Playback Switch", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), -WSS_DOUBLE("Synth Volume", 0, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1), +WSS_DOUBLE_TLV("Synth Volume", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1, + db_scale_5bit_12db_max), WSS_DOUBLE("Synth Capture Switch", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 6, 6, 1, 1), WSS_DOUBLE("Synth Capture Bypass", 0, @@ -776,14 +825,16 @@ CS4236_DOUBLE("Mic Playback Switch", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 6, 6, 1, 1), CS4236_DOUBLE("Mic Capture Switch", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 7, 7, 1, 1), -CS4236_DOUBLE("Mic Volume", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 0, 0, 31, 1), +CS4236_DOUBLE_TLV("Mic Volume", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, + 0, 0, 31, 1, db_scale_5bit_22db_max), CS4236_DOUBLE("Mic Playback Boost (+20dB)", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 5, 5, 1, 0), WSS_DOUBLE("Line Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Line Volume", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1), +WSS_DOUBLE_TLV("Line Volume", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1, + db_scale_5bit_12db_max), WSS_DOUBLE("Line Capture Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 6, 6, 1, 1), WSS_DOUBLE("Line Capture Bypass", 0, @@ -791,8 +842,9 @@ WSS_DOUBLE("Line Capture Bypass", 0, WSS_DOUBLE("CD Playback Switch", 0, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("CD Volume", 0, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), +WSS_DOUBLE_TLV("CD Volume", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1, + db_scale_5bit_12db_max), WSS_DOUBLE("CD Capture Switch", 0, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 6, 6, 1, 1), @@ -800,44 +852,53 @@ CS4236_DOUBLE1("Mono Output Playback Switch", 0, CS4231_MONO_CTRL, CS4236_RIGHT_MIX_CTRL, 6, 7, 1, 1), CS4236_DOUBLE1("Beep Playback Switch", 0, CS4231_MONO_CTRL, CS4236_LEFT_MIX_CTRL, 7, 7, 1, 1), -WSS_SINGLE("Beep Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), +WSS_SINGLE_TLV("Beep Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1, + db_scale_4bit), WSS_SINGLE("Beep Bypass Playback Switch", 0, CS4231_MONO_CTRL, 5, 1, 0), -WSS_DOUBLE("Capture Volume", 0, - CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), +WSS_DOUBLE_TLV("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, + 0, 0, 15, 0, db_scale_rec_gain), WSS_DOUBLE("Analog Loopback Capture Switch", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 7, 7, 1, 0), -WSS_SINGLE("Digital Loopback Playback Switch", 0, CS4231_LOOPBACK, 0, 1, 0), -CS4236_DOUBLE1("Digital Loopback Playback Volume", 0, - CS4231_LOOPBACK, CS4236_RIGHT_LOOPBACK, 2, 0, 63, 1) +WSS_SINGLE("Loopback Digital Playback Switch", 0, CS4231_LOOPBACK, 0, 1, 0), +CS4236_DOUBLE1_TLV("Loopback Digital Playback Volume", 0, + CS4231_LOOPBACK, CS4236_RIGHT_LOOPBACK, 2, 0, 63, 1, + db_scale_6bit), }; +static const DECLARE_TLV_DB_SCALE(db_scale_5bit_6db_max, -5600, 200, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_2bit_16db_max, -2400, 800, 0); + static struct snd_kcontrol_new snd_cs4235_controls[] = { WSS_DOUBLE("Master Playback Switch", 0, CS4235_LEFT_MASTER, CS4235_RIGHT_MASTER, 7, 7, 1, 1), -WSS_DOUBLE("Master Playback Volume", 0, - CS4235_LEFT_MASTER, CS4235_RIGHT_MASTER, 0, 0, 31, 1), +WSS_DOUBLE_TLV("Master Playback Volume", 0, + CS4235_LEFT_MASTER, CS4235_RIGHT_MASTER, 0, 0, 31, 1, + db_scale_5bit_6db_max), -CS4235_OUTPUT_ACCU("Playback Volume", 0), +CS4235_OUTPUT_ACCU("Playback Volume", 0, db_scale_2bit_16db_max), WSS_DOUBLE("Synth Playback Switch", 1, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), WSS_DOUBLE("Synth Capture Switch", 1, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 6, 6, 1, 1), -WSS_DOUBLE("Synth Volume", 1, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1), +WSS_DOUBLE_TLV("Synth Volume", 1, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1, + db_scale_5bit_12db_max), -CS4236_DOUBLE("Capture Volume", 0, - CS4236_LEFT_MIX_CTRL, CS4236_RIGHT_MIX_CTRL, 5, 5, 3, 1), +CS4236_DOUBLE_TLV("Capture Volume", 0, + CS4236_LEFT_MIX_CTRL, CS4236_RIGHT_MIX_CTRL, 5, 5, 3, 1, + db_scale_2bit), WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), WSS_DOUBLE("PCM Capture Switch", 0, CS4236_DAC_MUTE, CS4236_DAC_MUTE, 7, 6, 1, 1), -WSS_DOUBLE("PCM Volume", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1), +WSS_DOUBLE_TLV("PCM Volume", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1, + db_scale_6bit), CS4236_DOUBLE("DSP Switch", 0, CS4236_LEFT_DSP, CS4236_RIGHT_DSP, 7, 7, 1, 1), @@ -850,22 +911,25 @@ CS4236_DOUBLE("Mic Capture Switch", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 7, 7, 1, 1), CS4236_DOUBLE("Mic Playback Switch", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 6, 6, 1, 1), -CS4236_SINGLE("Mic Volume", 0, CS4236_LEFT_MIC, 0, 31, 1), +CS4236_SINGLE_TLV("Mic Volume", 0, CS4236_LEFT_MIC, 0, 31, 1, + db_scale_5bit_22db_max), CS4236_SINGLE("Mic Boost (+20dB)", 0, CS4236_LEFT_MIC, 5, 1, 0), WSS_DOUBLE("Line Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), WSS_DOUBLE("Line Capture Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 6, 6, 1, 1), -WSS_DOUBLE("Line Volume", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1), +WSS_DOUBLE_TLV("Line Volume", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1, + db_scale_5bit_12db_max), WSS_DOUBLE("CD Playback Switch", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), WSS_DOUBLE("CD Capture Switch", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 6, 6, 1, 1), -WSS_DOUBLE("CD Volume", 1, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), +WSS_DOUBLE_TLV("CD Volume", 1, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1, + db_scale_5bit_12db_max), CS4236_DOUBLE1("Beep Playback Switch", 0, CS4231_MONO_CTRL, CS4236_LEFT_MIX_CTRL, 7, 7, 1, 1), -- 1.6.4 ---------------------------------------------------------------------- Najlepsze oferty Sylwestrowe! Sprawdz >>> http://link.interia.pl/f2426

2 1

[alsa-devel] Patches for Samsung SoC PCM Controller driver
by Jassi Brar 19 Nov '09

19 Nov '09

Hello, (I re-submit patches after making changes suggested yesterday by Ben Dooks.) Samsung SoCs S3C2450, S3C64XX and later versions have got PCM controllers which drive what the ALSA stack terms DSP_A/DSP_B formats. Since the S3C ALSA platform driver was named s3c24xx-pcm, which cause direct namespace clash with the new PCM controller drivers, it has been renamed along with some of its local symbols. First two of the following series of patches does that and later ones make ground for and add the PCM controller driver. regards.

3 5

[alsa-devel] PATCH: ALSA - hda - Add support for Dell E6400 docking station
by Christian Hergert 19 Nov '09

19 Nov '09

Hi, I made a patch to fix bug 4657[1] on the tracker. I'm fairly certain that it isn't the correct approach but would like some feedback on what that would be. The power-mappings in patch_sigmatel.c seem to allow you to flip multiple bits. In the case of the Dell docking station, you need to enable some bits while disabling others. Therefore, instead of using the power mappings I just hacked in a quick check to work for the hardware I have on hand. If there is a proper way to fix this that I have missed, I'll be more than happy to fix the patch appropriately. [1] https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4657 Regards, -- Christian Hergert

2 5

[PATCH] Audio driver of Always Innovating OMAP3-based Touch Book
by Gregoire Gentil 19 Nov '09

19 Nov '09

Signed-off-by: Gregoire Gentil <gregoire(a)gentil.com> --- sound/soc/omap/Kconfig | 8 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/omap3touchbook.c | 211 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 221 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/omap3touchbook.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 653a362..c590252 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -83,6 +83,14 @@ config SND_OMAP_SOC_OMAP3_PANDORA help Say Y if you want to add support for SoC audio on the OMAP3 Pandora. +config SND_OMAP_SOC_OMAP3_TOUCHBOOK + tristate "SoC Audio support for OMAP3 Touch Book" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_TOUCHBOOK + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the OMAP3 Touch Book. + config SND_OMAP_SOC_OMAP3_BEAGLE tristate "SoC Audio support for OMAP3 Beagle" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_BEAGLE diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 02d6947..f360b82 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -15,6 +15,7 @@ snd-soc-omap3evm-objs := omap3evm.o snd-soc-sdp3430-objs := sdp3430.o snd-soc-omap3pandora-objs := omap3pandora.o snd-soc-omap3beagle-objs := omap3beagle.o +snd-soc-omap3touchbook-objs := omap3touchbook.o snd-soc-zoom2-objs := zoom2.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o @@ -26,4 +27,5 @@ obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(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_OMAP3_TOUCHBOOK) += snd-soc-omap3touchbook.o obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o diff --git a/sound/soc/omap/omap3touchbook.c b/sound/soc/omap/omap3touchbook.c new file mode 100644 index 0000000..584f07d --- /dev/null +++ b/sound/soc/omap/omap3touchbook.c @@ -0,0 +1,211 @@ +/* + * omap3touchbook.c -- SoC audio for OMAP3 Touch Book + * + * Author: Gregoire Gentil <gregoire(a)gentil.com> + * + * Derived from sound/soc/omap/omap3beagle.c + * + * 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/irq.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <plat/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +#define TB_HEADPHONE_GPIO 56 +#define TB_HEADPHONE_IRQ OMAP_GPIO_IRQ(TB_HEADPHONE_GPIO) + +static int omap3touchbook_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int fmt; + int ret; + + switch (params_channels(params)) { + case 2: /* Stereo I2S mode */ + fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + break; + case 4: /* Four channel TDM mode */ + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM; + break; + default: + return -EINVAL; + } + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops omap3touchbook_ops = { + .hw_params = omap3touchbook_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link omap3touchbook_dai = { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], + .ops = &omap3touchbook_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_omap3touchbook = { + .name = "omap3touchbook", + .platform = &omap_soc_platform, + .dai_link = &omap3touchbook_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device omap3touchbook_snd_devdata = { + .card = &snd_soc_omap3touchbook, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *omap3touchbook_snd_device; + +static void jack_work_func(struct work_struct *wq) +{ + struct snd_soc_device *socdev = platform_get_drvdata( + omap3touchbook_snd_device); + struct snd_soc_codec *codec = socdev->card->codec; + + snd_soc_dapm_sync(codec); +} +DECLARE_WORK(jack_work, jack_work_func); + +static irqreturn_t touchbook_headphone_event(int irq, void *snd) +{ + int status = gpio_get_value(TB_HEADPHONE_GPIO); + struct snd_soc_device *socdev = platform_get_drvdata( + omap3touchbook_snd_device); + struct snd_soc_codec *codec = socdev->card->codec; + + if (status) { + snd_soc_dapm_disable_pin(codec, "HFL"); + snd_soc_dapm_disable_pin(codec, "HFR"); + } else { + snd_soc_dapm_enable_pin(codec, "HFL"); + snd_soc_dapm_enable_pin(codec, "HFR"); + } + + schedule_work(&jack_work); + return IRQ_HANDLED; +} + +static int __init omap3touchbook_soc_init(void) +{ + int ret; + + if (!machine_is_touchbook()) { + pr_debug("Not OMAP3 Touch Book!\n"); + return -ENODEV; + } + pr_info("OMAP3 Touch Book SoC init\n"); + + omap3touchbook_snd_device = platform_device_alloc("soc-audio", -1); + if (!omap3touchbook_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(omap3touchbook_snd_device, + &omap3touchbook_snd_devdata); + omap3touchbook_snd_devdata.dev = &omap3touchbook_snd_device->dev; + *(unsigned int *)omap3touchbook_dai.cpu_dai->private_data = 1; + + /* Touch Book -- headphone jack sensor */ + omap_set_gpio_debounce(TB_HEADPHONE_GPIO, 1); + omap_set_gpio_debounce_time(TB_HEADPHONE_GPIO, 0xff); + + ret = request_irq(TB_HEADPHONE_IRQ, touchbook_headphone_event, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING, + "touchbook_headphone", + omap3touchbook_snd_device); + if (ret < 0) + goto err1; + + ret = platform_device_add(omap3touchbook_snd_device); + if (ret) + goto err1; + + /* Detect headphone status */ + touchbook_headphone_event(0, omap3touchbook_snd_device); + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(omap3touchbook_snd_device); + + return ret; +} + +static void __exit omap3touchbook_soc_exit(void) +{ + platform_device_unregister(omap3touchbook_snd_device); +} + +module_init(omap3touchbook_soc_init); +module_exit(omap3touchbook_soc_exit); + +MODULE_AUTHOR("Gregoire Gentil <gregoire(a)gentil.com>"); +MODULE_DESCRIPTION("ALSA SoC OMAP3 Touch Book"); +MODULE_LICENSE("GPL"); -- 1.6.0.4 --=-PeKNApWMNzEFfot32AT+ Content-Disposition: attachment; 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1 0

[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 18 Nov '09

18 Nov '09

The following changes since commit f9ede4eca01cc64ce37549c282b6fde727c0ec84: Marin Mitov (1): ASoC: Use DMA_BIT_MASK(32) instead of deprecated DMA_32BIT_MASK are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Anuj Aggarwal (1): ASoC: Modifying the license string GPLv2 for OMAP3 EVM Grazvydas Ignotas (1): ASoC: OMAP3 Pandora: update for TWL4030 codec changes Jassi Brar (6): ASoC: Rename s3c24xx_pcm prefix to s3c_dma ASoC: Rename 's3c24xx-pcm' driver to 's3c-dma' ARM: S3C64XX: Define PCM Controller base registers ARM: S3C64XX: Added platform data header ARM: S3C64XX: Defined PCM controller platform devices ASoC: Added the CPU driver for PCM controllers Mark Brown (3): ASoC: Remove redundant snd_soc_dapm_new_widgets() calls Merge branch 'topic/asoc' of git://git.kernel.org/.../tiwai/sound-2.6 into for-2.6.33 Merge branch 'for-2.6.32' into for-2.6.33 Mike Rapoport (1): ASoC: OMAP: enable Overo driver for CM-T35 Troy Kisky (2): ASoC: DaVinci: remove requirement that dma_params is 1st in structure ASoC: tlv320aic23 fix rate selection arch/arm/mach-s3c6400/include/mach/map.h | 2 + arch/arm/plat-s3c/include/plat/audio.h | 17 + arch/arm/plat-s3c/include/plat/devs.h | 3 + arch/arm/plat-s3c64xx/dev-audio.c | 101 +++++- sound/soc/codecs/ad1836.c | 1 - sound/soc/codecs/ad1938.c | 1 - sound/soc/codecs/ak4535.c | 1 - sound/soc/codecs/ak4671.c | 1 - sound/soc/codecs/cx20442.c | 1 - sound/soc/codecs/ssm2602.c | 1 - sound/soc/codecs/tlv320aic23.c | 3 +- sound/soc/codecs/tlv320aic3x.c | 1 - sound/soc/codecs/tlv320dac33.c | 1 - sound/soc/codecs/twl4030.c | 1 - sound/soc/codecs/uda1380.c | 1 - sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8400.c | 1 - sound/soc/codecs/wm8510.c | 1 - sound/soc/codecs/wm8523.c | 1 - sound/soc/codecs/wm8580.c | 1 - sound/soc/codecs/wm8711.c | 1 - sound/soc/codecs/wm8728.c | 2 - sound/soc/codecs/wm8731.c | 1 - sound/soc/codecs/wm8750.c | 1 - sound/soc/codecs/wm8753.c | 1 - sound/soc/codecs/wm8900.c | 2 - sound/soc/codecs/wm8903.c | 2 - sound/soc/codecs/wm8940.c | 1 - sound/soc/codecs/wm8960.c | 1 - sound/soc/codecs/wm8961.c | 1 - sound/soc/codecs/wm8971.c | 2 - sound/soc/codecs/wm8974.c | 1 - sound/soc/codecs/wm8988.c | 1 - sound/soc/codecs/wm8990.c | 1 - sound/soc/codecs/wm8993.c | 2 - sound/soc/codecs/wm9081.c | 1 - sound/soc/codecs/wm9705.c | 1 - sound/soc/codecs/wm9712.c | 1 - sound/soc/codecs/wm9713.c | 1 - sound/soc/davinci/davinci-i2s.c | 6 +- sound/soc/davinci/davinci-mcasp.c | 1 + sound/soc/davinci/davinci-mcasp.h | 5 - sound/soc/davinci/davinci-pcm.c | 7 +- sound/soc/omap/Kconfig | 7 +- sound/soc/omap/omap3evm.c | 2 +- sound/soc/omap/omap3pandora.c | 3 +- sound/soc/omap/overo.c | 4 +- sound/soc/s3c24xx/Kconfig | 3 + sound/soc/s3c24xx/Makefile | 4 +- sound/soc/s3c24xx/jive_wm8750.c | 2 +- sound/soc/s3c24xx/ln2440sbc_alc650.c | 2 +- sound/soc/s3c24xx/neo1973_gta02_wm8753.c | 2 +- sound/soc/s3c24xx/neo1973_wm8753.c | 2 +- sound/soc/s3c24xx/{s3c24xx-pcm.c => s3c-dma.c} | 87 ++-- sound/soc/s3c24xx/{s3c24xx-pcm.h => s3c-dma.h} | 8 +- sound/soc/s3c24xx/s3c-i2s-v2.c | 4 +- sound/soc/s3c24xx/s3c-i2s-v2.h | 4 +- sound/soc/s3c24xx/s3c-pcm.c | 552 ++++++++++++++++++++++++ sound/soc/s3c24xx/s3c-pcm.h | 123 ++++++ sound/soc/s3c24xx/s3c2412-i2s.c | 6 +- sound/soc/s3c24xx/s3c2443-ac97.c | 12 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 12 +- sound/soc/s3c24xx/s3c24xx_simtec.c | 2 +- sound/soc/s3c24xx/s3c24xx_simtec_hermes.c | 2 +- sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c | 2 +- sound/soc/s3c24xx/s3c24xx_uda134x.c | 2 +- sound/soc/s3c24xx/s3c64xx-i2s.c | 6 +- sound/soc/s3c24xx/smdk2443_wm9710.c | 2 +- sound/soc/s3c24xx/smdk64xx_wm8580.c | 2 +- 69 files changed, 900 insertions(+), 142 deletions(-) create mode 100644 arch/arm/plat-s3c/include/plat/audio.h rename sound/soc/s3c24xx/{s3c24xx-pcm.c => s3c-dma.c} (82%) rename sound/soc/s3c24xx/{s3c24xx-pcm.h => s3c-dma.h} (87%) create mode 100644 sound/soc/s3c24xx/s3c-pcm.c create mode 100644 sound/soc/s3c24xx/s3c-pcm.h

2 1

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 18 Nov '09

18 Nov '09

The following changes since commit 6fc786d5034ed7ce2d43c459211137de6d99dd28: Jassi Brar (1): ASoC: S3C64XX I2S: Enable audio-bus clock are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 All very driver-local so should be low impact. Anuj Aggarwal (1): ASoC: Modifying the license string GPLv2 for OMAP3 EVM Grazvydas Ignotas (1): ASoC: OMAP3 Pandora: update for TWL4030 codec changes Troy Kisky (1): ASoC: tlv320aic23 fix rate selection sound/soc/codecs/tlv320aic23.c | 2 +- sound/soc/omap/omap3evm.c | 2 +- sound/soc/omap/omap3pandora.c | 3 ++- 3 files changed, 4 insertions(+), 3 deletions(-)

2 1

[alsa-devel] [PATCH 1/2] ASoC: Correcting the config options in soc/omap/Makefile
by Anuj Aggarwal 18 Nov '09

18 Nov '09

Wrong config options were being used in the soc/omap/Makefile for OMAP2 & OMAP3 and AM3517 EVMs. Signed-off-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> --- sound/soc/omap/Makefile | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 02d6947..a69bd04 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -21,8 +21,8 @@ obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(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 -obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o -obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o +obj-$(CONFIG_SND_SOC_OMAP_OMAP3EVM) += snd-soc-omap3evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o -- 1.6.2.4

2 1

[alsa-devel] patch_realtek: support Vaio Type G
by Mattia Dongili 18 Nov '09

18 Nov '09

Hi Takashi, my laptop (Vaio Type G) used to work (although the valume was pretty low) with model=hippo until 2.6.29 then from 2.6.30 it stopped being supported throwing this message: hda-intel: azx_get_response timeout, switching to polling mode: last cmd=0x01570900 I recently found this rather old post[1] that has a patch to patch_realtek.c that adds an additional model and actually makes my soundcard happy again. I forward ported to the current kernel (2.6.32-rc7) and renamed variables/functions a little bit to make it less specific. There might be better way of supporting my soundcard anyway this is one as far as I can say. I attached the output of alsa-info.sh just in case. Not sure how signing-off or attribution could work in this case and honestly my japanese language is quite weak to be able to contact the author. If you want I can send a properly signed-off version, in any case it would be great if you could merge it and push it to Linus. Thanks [1]: http://blog.livedoor.jp/yoyo_real/archives/51872668.html --- diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ff20048..d47f5f5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -99,6 +99,7 @@ enum { ALC262_HP_RP5700, ALC262_BENQ_ED8, ALC262_SONY_ASSAMD, + ALC262_SONY_VGN_G, ALC262_BENQ_T31, ALC262_ULTRA, ALC262_LENOVO_3000, @@ -10329,6 +10330,17 @@ static struct snd_kcontrol_new alc262_sony_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc262_sony_vgn_g_mixer[] = { + HDA_CODEC_VOLUME_MONO("Front Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Front Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + { } /* end */ +}; + static struct snd_kcontrol_new alc262_benq_t31_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), ALC262_HIPPO_MASTER_SWITCH, @@ -10459,7 +10471,6 @@ static struct hda_verb alc262_init_verbs[] = { {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, - { } }; @@ -10489,6 +10500,52 @@ static struct hda_verb alc262_sony_unsol_verbs[] = { {} }; +static struct hda_verb alc262_sony_vgn_g_unsol_verbs[] = { + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0}, + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {} +}; + +static struct hda_input_mux alc262_sony_vgn_g_capture_source = { + .num_items = 1, + .items = { + { "Mic", 0x0 }, + }, +}; + +/* mute/unmute sony VGN-G 0x16 internal speaker 0x15 headphone */ +static void alc262_sony_vgn_g_automute(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + unsigned int mute; + unsigned int present; + + /* need to execute and sync at first */ + snd_hda_codec_read(codec, 0x15, 0, AC_VERB_SET_PIN_SENSE, 0); + present = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0); + spec->jack_present = (present & 0x80000000) != 0; + if (spec->jack_present) { + /* mute internal speaker */ + snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, + HDA_AMP_MUTE, HDA_AMP_MUTE); + } else { + /* unmute internal speaker if necessary */ + mute = snd_hda_codec_amp_read(codec, 0x15, 0, HDA_OUTPUT, 0); + snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); + } +} + +static void alc262_sony_vgn_g_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) != ALC880_HP_EVENT) + return; + alc262_sony_vgn_g_automute(codec); +} + static struct snd_kcontrol_new alc262_toshiba_s06_mixer[] = { HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), @@ -11423,6 +11480,7 @@ static const char *alc262_models[ALC262_MODEL_LAST] = { [ALC262_BENQ_ED8] = "benq", [ALC262_BENQ_T31] = "benq-t31", [ALC262_SONY_ASSAMD] = "sony-assamd", + [ALC262_SONY_VGN_G] = "sony-vgn-g", [ALC262_TOSHIBA_S06] = "toshiba-s06", [ALC262_TOSHIBA_RX1] = "toshiba-rx1", [ALC262_ULTRA] = "ultra", @@ -11456,6 +11514,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { ALC262_HP_TC_T5735), SND_PCI_QUIRK(0x103c, 0x2817, "HP RP5700", ALC262_HP_RP5700), SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD), + SND_PCI_QUIRK(0x104d, 0x8208, "Sony VGN-G", ALC262_SONY_VGN_G), SND_PCI_QUIRK(0x104d, 0x8203, "Sony UX-90", ALC262_HIPPO), SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */ @@ -11613,6 +11672,18 @@ static struct alc_config_preset alc262_presets[] = { .setup = alc262_hippo_setup, .init_hook = alc262_hippo_automute, }, + [ALC262_SONY_VGN_G] = { + .mixers = { alc262_sony_vgn_g_mixer }, + .init_verbs = { alc262_init_verbs, alc262_sony_vgn_g_unsol_verbs }, + .num_dacs = ARRAY_SIZE(alc262_dac_nids), + .dac_nids = alc262_dac_nids, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc262_modes), + .channel_mode = alc262_modes, + .input_mux = &alc262_sony_vgn_g_capture_source, + .unsol_event = alc262_sony_vgn_g_unsol_event, + .init_hook = alc262_sony_vgn_g_automute, + }, [ALC262_BENQ_T31] = { .mixers = { alc262_benq_t31_mixer }, .init_verbs = { alc262_init_verbs, alc262_benq_t31_EAPD_verbs, -- mattia :wq!

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Re: [alsa-devel] Artifacts present in AIC23 capture for 48 KHz sampling rate
by Aggarwal, Anuj 18 Nov '09

18 Nov '09

> -----Original Message----- > From: Aggarwal, Anuj > Sent: Friday, November 06, 2009 6:28 PM > To: alsa-devel(a)alsa-project.org; 'linux-omap(a)vger.kernel.org' > Subject: Artifacts present in AIC23 capture for 48 KHz sampling rate > > Hi, > > I am observing artifacts (sharp spikes at fixed intervals) while > capturing audio on AM3517 EVM and AIC23 codec. They are present only > in one of the channels when I am capturing at 48 KHz. All other > sampling rates are working fine with the above said combination. > > I have also attached the screenshot taken with the help of Audacity > utility. Here, I tried recording silence but artifacts were observed > on one channel. > > Has anyone also observed the similar behavior with AIC23 codec? Any > hints on what could be the root cause? [Aggarwal, Anuj] On further debugging, I found that the function find_rate() in sound/soc/codecs/tlv320aic23.c is not returning the correct value for capture in 48KHz sample rate. In USB mode (MCLK=12MHz), for 48KHz, it returns 0x7D (CLKOUT=0, CLKIN=1, SR[3:0]=0xF, BOSR=1, Normal=1) whereas as per the AIC23B spec, it should have been 0x5D (CLKOUT=0, CLKIN=1, SR[3:0]=0x7, BOSR=1, Normal=1). When I forcefully write the above said value to the Sample Rate Control register, things work fine for me for 48KHz capture. Is my understanding correct for this problem? Can someone help me understand how the function calculates the appropriate value? > > Regards, > Anuj Aggarwal

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Re: [alsa-devel] [PATCH (for-next 2)] ASoC: Add support for IGEP v2
by Mike Rapoport 18 Nov '09

18 Nov '09

(adding ALSA mailing list to CC again :) ) Enric Balletbo i Serra wrote: > From: Enric Balletbo i Serra <eballetbo(a)iseebcn.com> > > > Signed-off-by: Enric Balletbo i Serra <eballetbo(a)iseebcn.com> > --- > sound/soc/omap/Kconfig | 7 ++ > sound/soc/omap/Makefile | 2 + > sound/soc/omap/igep0020.c | 148 +++++++++++++++++++++++++++++++++++++++++++++ > 3 files changed, 157 insertions(+), 0 deletions(-) > create mode 100644 sound/soc/omap/igep0020.c > > diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig > index 653a362..5cc8a63 100644 > --- a/sound/soc/omap/Kconfig > +++ b/sound/soc/omap/Kconfig > @@ -99,3 +99,10 @@ config SND_OMAP_SOC_ZOOM2 > help > Say Y if you want to add support for Soc audio on Zoom2 board. > > +config SND_OMAP_SOC_IGEP0020 > + tristate "SoC Audio support for IGEP v2" > + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_IGEP0020 > + select SND_OMAP_SOC_MCBSP > + select SND_SOC_TWL4030 > + help > + Say Y if you want to add support for Soc audio on IGEP v2 board. > diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile > index 02d6947..d4ed23a 100644 > --- a/sound/soc/omap/Makefile > +++ b/sound/soc/omap/Makefile > @@ -16,6 +16,7 @@ snd-soc-sdp3430-objs := sdp3430.o > snd-soc-omap3pandora-objs := omap3pandora.o > snd-soc-omap3beagle-objs := omap3beagle.o > 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_AMS_DELTA) += snd-soc-ams-delta.o > @@ -27,3 +28,4 @@ obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o > obj-$(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 > +obj-$(CONFIG_SND_OMAP_SOC_IGEP0020) += snd-soc-igep0020.o > diff --git a/sound/soc/omap/igep0020.c b/sound/soc/omap/igep0020.c > new file mode 100644 > index 0000000..3583c42 > --- /dev/null > +++ b/sound/soc/omap/igep0020.c > @@ -0,0 +1,148 @@ > +/* > + * igep0020.c -- SoC audio for IGEP v2 > + * > + * Based on sound/soc/omap/overo.c by Steve Sakoman > + * > + * This program is free software; you can redistribute it and/or > + * modify it under the terms of the GNU General Public License > + * version 2 as published by the Free Software Foundation. > + * > + * This program is distributed in the hope that it will be useful, but > + * WITHOUT ANY WARRANTY; without even the implied warranty of > + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU > + * General Public License for more details. > + * > + * You should have received a copy of the GNU General Public License > + * along with this program; if not, write to the Free Software > + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA > + * 02110-1301 USA > + * > + */ > + > +#include <linux/clk.h> > +#include <linux/platform_device.h> > +#include <sound/core.h> > +#include <sound/pcm.h> > +#include <sound/soc.h> > +#include <sound/soc-dapm.h> > + > +#include <asm/mach-types.h> > +#include <mach/hardware.h> > +#include <mach/gpio.h> > +#include <plat/mcbsp.h> > + > +#include "omap-mcbsp.h" > +#include "omap-pcm.h" > +#include "../codecs/twl4030.h" > + > +static int igep2_hw_params(struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *params) > +{ > + struct snd_soc_pcm_runtime *rtd = substream->private_data; > + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; > + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; > + int ret; > + > + /* Set codec DAI configuration */ > + ret = snd_soc_dai_set_fmt(codec_dai, > + SND_SOC_DAIFMT_I2S | > + SND_SOC_DAIFMT_NB_NF | > + SND_SOC_DAIFMT_CBM_CFM); > + if (ret < 0) { > + printk(KERN_ERR "can't set codec DAI configuration\n"); > + return ret; > + } > + > + /* Set cpu DAI configuration */ > + ret = snd_soc_dai_set_fmt(cpu_dai, > + SND_SOC_DAIFMT_I2S | > + SND_SOC_DAIFMT_NB_NF | > + SND_SOC_DAIFMT_CBM_CFM); > + if (ret < 0) { > + printk(KERN_ERR "can't set cpu DAI configuration\n"); > + return ret; > + } > + > + /* Set the codec system clock for DAC and ADC */ > + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, > + SND_SOC_CLOCK_IN); > + if (ret < 0) { > + printk(KERN_ERR "can't set codec system clock\n"); > + return ret; > + } > + > + return 0; > +} > + > +static struct snd_soc_ops igep2_ops = { > + .hw_params = igep2_hw_params, > +}; > + > +/* Digital audio interface glue - connects codec <--> CPU */ > +static struct snd_soc_dai_link igep2_dai = { > + .name = "TWL4030", > + .stream_name = "TWL4030", > + .cpu_dai = &omap_mcbsp_dai[0], > + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], > + .ops = &igep2_ops, > +}; > + > +/* Audio machine driver */ > +static struct snd_soc_card snd_soc_card_igep2 = { > + .name = "igep2", > + .platform = &omap_soc_platform, > + .dai_link = &igep2_dai, > + .num_links = 1, > +}; > + > +/* Audio subsystem */ > +static struct snd_soc_device igep2_snd_devdata = { > + .card = &snd_soc_card_igep2, > + .codec_dev = &soc_codec_dev_twl4030, > +}; > + > +static struct platform_device *igep2_snd_device; > + > +static int __init igep2_soc_init(void) > +{ > + int ret; > + > + if (!machine_is_igep0020()) { > + pr_debug("Not IGEP v2!\n"); > + return -ENODEV; > + } > + printk(KERN_INFO "IGEP v2 SoC init\n"); > + > + igep2_snd_device = platform_device_alloc("soc-audio", -1); > + if (!igep2_snd_device) { > + printk(KERN_ERR "Platform device allocation failed\n"); > + return -ENOMEM; > + } > + > + platform_set_drvdata(igep2_snd_device, &igep2_snd_devdata); > + igep2_snd_devdata.dev = &igep2_snd_device->dev; > + *(unsigned int *)igep2_dai.cpu_dai->private_data = 1; /* McBSP2 */ > + > + ret = platform_device_add(igep2_snd_device); > + if (ret) > + goto err1; > + > + return 0; > + > +err1: > + printk(KERN_ERR "Unable to add platform device\n"); > + platform_device_put(igep2_snd_device); > + > + return ret; > +} > +module_init(igep2_soc_init); > + > +static void __exit igep2_soc_exit(void) > +{ > + platform_device_unregister(igep2_snd_device); > +} > +module_exit(igep2_soc_exit); > + > +MODULE_AUTHOR("Enric Balletbo i Serra <eballetbo(a)iseebcn.com>"); > +MODULE_DESCRIPTION("ALSA SoC IGEP v2"); > +MODULE_LICENSE("GPL"); -- Sincerely yours, Mike.

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[alsa-devel] [PATCH] ASoC: OMAP3 Pandora: update for TWL4030 codec changes
by Grazvydas Ignotas 18 Nov '09

18 Nov '09

A while ago TWL4030 had it's playback stream name changed, but pandora needs it for it's playback path. Update to correct stream name so that playback works again. Also mark VIBRA output as not connected. Signed-off-by: Grazvydas Ignotas <notasas(a)gmail.com> --- sound/soc/omap/omap3pandora.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index ad219aa..0cd06f5 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -134,7 +134,7 @@ static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w, * |P| <--- TWL4030 <--------- Line In and MICs */ static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = { - SND_SOC_DAPM_DAC("PCM DAC", "Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("PCM DAC", "HiFi Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_PGA_E("Headphone Amplifier", SND_SOC_NOPM, 0, 0, NULL, 0, omap3pandora_hp_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), @@ -181,6 +181,7 @@ static int omap3pandora_out_init(struct snd_soc_codec *codec) snd_soc_dapm_nc_pin(codec, "CARKITR"); snd_soc_dapm_nc_pin(codec, "HFL"); snd_soc_dapm_nc_pin(codec, "HFR"); + snd_soc_dapm_nc_pin(codec, "VIBRA"); ret = snd_soc_dapm_new_controls(codec, omap3pandora_out_dapm_widgets, ARRAY_SIZE(omap3pandora_out_dapm_widgets)); -- 1.5.6.3

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[alsa-devel] [PATCH 2/2] ASoC: Modifying the license string GPLv2 for OMAP3 EVM
by Anuj Aggarwal 18 Nov '09

18 Nov '09

Correcting the license string from GPLv2 -> GPL v2. Found the problem while building OMAP3 ASoC driver as module. Signed-off-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> --- sound/soc/omap/omap3evm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c index 41a91b5..627fadb 100644 --- a/sound/soc/omap/omap3evm.c +++ b/sound/soc/omap/omap3evm.c @@ -144,4 +144,4 @@ module_exit(omap3evm_soc_exit); MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal(a)ti.com>"); MODULE_DESCRIPTION("ALSA SoC OMAP3 EVM"); -MODULE_LICENSE("GPLv2"); +MODULE_LICENSE("GPL v2"); -- 1.6.2.4

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[alsa-devel] Status of ALSA support for MacBook Pro 5, 2 in kernel 2.6.32
by Adolf Winterer 18 Nov '09

18 Nov '09

Hello, there had been a lot of messages here on the list and in other internet sources about patches to snd-hda-intel regarding the sound support for MacBook Pro 5,1 and MacBook Pro 5,2. But the situation is far from being clear. Some sources stated that the patches for the MacBook Pro 5,1 would support the MacBook Pro 5,2 as well, others disagreed. Recently on this list I've seen efforts to integrate the current ALSA 1.0.21+x into kernel 2.6.32. Fact is that with kernel 2.6.30-2, which has ALSA 1.0.20, from Debian testing my MacBook Pro 5,2 is not working. There is no sound at all. Compilation of ALSA sources 1.0.21 using the module assistant fails with an error message regarding a missing curly bracket. There is a bug report in the Debian bug tracker regarding this, but no progress is visible. Compiling the sources directly using the 1.0.21 sources from the ALSA project page goes through, but the resulting modules do not load, give an error message about an unknown symbol. Therefore I was not able to find out if ALSA 1.0.21 supports the MacBook Pro 5,2. Please, can anyone of the development team help with a statement clearing up things by answering the following questions? * Will the support for the MacBook Pro 5,2 be part of the soon to be released kernel 2.6.32? * Or is it already present in kernel 2.6.31? * What is the correct parameter for the loading of the module? Is it just "auto" or "mb5"? Or anything else? Your help is greatly appreciated. Best regards, Adolf

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[alsa-devel] [PATCH 00/10] Intel HDMI fixes v2
by Wu Fengguang 18 Nov '09

18 Nov '09

Takashi, Here are the updated Intel HDMI fixes. Thanks, Fengguang

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Re: [alsa-devel] [PATCH (for-next)] ASoC: Add support for IGEP v2
by Mike Rapoport 18 Nov '09

18 Nov '09

(adding ALSA list to CC) Enric Balletbo i Serra wrote: > From: Enric Balletbo i Serra <eballetbo(a)iseebcn.com> > > > Signed-off-by: Enric Balletbo i Serra <eballetbo(a)iseebcn.com> > --- > arch/arm/mach-omap2/Kconfig | 2 +- > sound/soc/omap/Kconfig | 7 ++ > sound/soc/omap/Makefile | 2 + > sound/soc/omap/igep0020.c | 148 +++++++++++++++++++++++++++++++++++++++++++ > 4 files changed, 158 insertions(+), 1 deletions(-) > create mode 100644 sound/soc/omap/igep0020.c > > diff --git a/arch/arm/mach-omap2/Kconfig b/arch/arm/mach-omap2/Kconfig > index 68866f3..6922198 100644 > --- a/arch/arm/mach-omap2/Kconfig > +++ b/arch/arm/mach-omap2/Kconfig > @@ -114,7 +114,7 @@ config MACH_OMAP_3630SDP > depends on ARCH_OMAP3 && ARCH_OMAP34XX > > config MACH_IGEP0020 > - bool "IGEP0020" > + bool "IGEP v2 board" > depends on ARCH_OMAP3 && ARCH_OMAP34XX > > config MACH_OMAP_4430SDP > diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig > index 653a362..5cc8a63 100644 > --- a/sound/soc/omap/Kconfig > +++ b/sound/soc/omap/Kconfig > @@ -99,3 +99,10 @@ config SND_OMAP_SOC_ZOOM2 > help > Say Y if you want to add support for Soc audio on Zoom2 board. > > +config SND_OMAP_SOC_IGEP0020 > + tristate "SoC Audio support for IGEP v2" > + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_IGEP0020 > + select SND_OMAP_SOC_MCBSP > + select SND_SOC_TWL4030 > + help > + Say Y if you want to add support for Soc audio on IGEP v2 board. > diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile > index 02d6947..d4ed23a 100644 > --- a/sound/soc/omap/Makefile > +++ b/sound/soc/omap/Makefile > @@ -16,6 +16,7 @@ snd-soc-sdp3430-objs := sdp3430.o > snd-soc-omap3pandora-objs := omap3pandora.o > snd-soc-omap3beagle-objs := omap3beagle.o > 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_AMS_DELTA) += snd-soc-ams-delta.o > @@ -27,3 +28,4 @@ obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o > obj-$(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 > +obj-$(CONFIG_SND_OMAP_SOC_IGEP0020) += snd-soc-igep0020.o > diff --git a/sound/soc/omap/igep0020.c b/sound/soc/omap/igep0020.c > new file mode 100644 > index 0000000..3583c42 > --- /dev/null > +++ b/sound/soc/omap/igep0020.c > @@ -0,0 +1,148 @@ > +/* > + * igep0020.c -- SoC audio for IGEP v2 > + * > + * Based on sound/soc/omap/overo.c by Steve Sakoman > + * > + * This program is free software; you can redistribute it and/or > + * modify it under the terms of the GNU General Public License > + * version 2 as published by the Free Software Foundation. > + * > + * This program is distributed in the hope that it will be useful, but > + * WITHOUT ANY WARRANTY; without even the implied warranty of > + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU > + * General Public License for more details. > + * > + * You should have received a copy of the GNU General Public License > + * along with this program; if not, write to the Free Software > + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA > + * 02110-1301 USA > + * > + */ > + > +#include <linux/clk.h> > +#include <linux/platform_device.h> > +#include <sound/core.h> > +#include <sound/pcm.h> > +#include <sound/soc.h> > +#include <sound/soc-dapm.h> > + > +#include <asm/mach-types.h> > +#include <mach/hardware.h> > +#include <mach/gpio.h> > +#include <plat/mcbsp.h> > + > +#include "omap-mcbsp.h" > +#include "omap-pcm.h" > +#include "../codecs/twl4030.h" > + > +static int igep2_hw_params(struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *params) > +{ > + struct snd_soc_pcm_runtime *rtd = substream->private_data; > + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; > + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; > + int ret; > + > + /* Set codec DAI configuration */ > + ret = snd_soc_dai_set_fmt(codec_dai, > + SND_SOC_DAIFMT_I2S | > + SND_SOC_DAIFMT_NB_NF | > + SND_SOC_DAIFMT_CBM_CFM); > + if (ret < 0) { > + printk(KERN_ERR "can't set codec DAI configuration\n"); > + return ret; > + } > + > + /* Set cpu DAI configuration */ > + ret = snd_soc_dai_set_fmt(cpu_dai, > + SND_SOC_DAIFMT_I2S | > + SND_SOC_DAIFMT_NB_NF | > + SND_SOC_DAIFMT_CBM_CFM); > + if (ret < 0) { > + printk(KERN_ERR "can't set cpu DAI configuration\n"); > + return ret; > + } > + > + /* Set the codec system clock for DAC and ADC */ > + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, > + SND_SOC_CLOCK_IN); > + if (ret < 0) { > + printk(KERN_ERR "can't set codec system clock\n"); > + return ret; > + } > + > + return 0; > +} > + > +static struct snd_soc_ops igep2_ops = { > + .hw_params = igep2_hw_params, > +}; > + > +/* Digital audio interface glue - connects codec <--> CPU */ > +static struct snd_soc_dai_link igep2_dai = { > + .name = "TWL4030", > + .stream_name = "TWL4030", > + .cpu_dai = &omap_mcbsp_dai[0], > + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], > + .ops = &igep2_ops, > +}; > + > +/* Audio machine driver */ > +static struct snd_soc_card snd_soc_card_igep2 = { > + .name = "igep2", > + .platform = &omap_soc_platform, > + .dai_link = &igep2_dai, > + .num_links = 1, > +}; > + > +/* Audio subsystem */ > +static struct snd_soc_device igep2_snd_devdata = { > + .card = &snd_soc_card_igep2, > + .codec_dev = &soc_codec_dev_twl4030, > +}; > + > +static struct platform_device *igep2_snd_device; > + > +static int __init igep2_soc_init(void) > +{ > + int ret; > + > + if (!machine_is_igep0020()) { > + pr_debug("Not IGEP v2!\n"); > + return -ENODEV; > + } > + printk(KERN_INFO "IGEP v2 SoC init\n"); > + > + igep2_snd_device = platform_device_alloc("soc-audio", -1); > + if (!igep2_snd_device) { > + printk(KERN_ERR "Platform device allocation failed\n"); > + return -ENOMEM; > + } > + > + platform_set_drvdata(igep2_snd_device, &igep2_snd_devdata); > + igep2_snd_devdata.dev = &igep2_snd_device->dev; > + *(unsigned int *)igep2_dai.cpu_dai->private_data = 1; /* McBSP2 */ > + > + ret = platform_device_add(igep2_snd_device); > + if (ret) > + goto err1; > + > + return 0; > + > +err1: > + printk(KERN_ERR "Unable to add platform device\n"); > + platform_device_put(igep2_snd_device); > + > + return ret; > +} > +module_init(igep2_soc_init); > + > +static void __exit igep2_soc_exit(void) > +{ > + platform_device_unregister(igep2_snd_device); > +} > +module_exit(igep2_soc_exit); > + > +MODULE_AUTHOR("Enric Balletbo i Serra <eballetbo(a)iseebcn.com>"); > +MODULE_DESCRIPTION("ALSA SoC IGEP v2"); > +MODULE_LICENSE("GPL"); -- Sincerely yours, Mike.

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[alsa-devel] [patch] remove unnecessary null check
by Dan Carpenter 18 Nov '09

18 Nov '09

This function is only called from snd_ctl_ioctl() and the file parameter can never be null so there is no need to check it here. We dereference file at the start of the function: struct snd_card *card = file->card; and it confuses static checkers to dereference a pointer before checking it. regards, dan carpenter Signed-off-by: Dan Carpenter <error27(a)gmail.com> --- orig/sound/core/control.c 2009-11-17 07:15:04.000000000 +0200 +++ devel/sound/core/control.c 2009-11-17 07:40:30.000000000 +0200 @@ -1120,7 +1120,7 @@ goto __kctl_end; } if (vd->access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { - if (file && vd->owner != NULL && vd->owner != file) { + if (vd->owner != NULL && vd->owner != file) { err = -EPERM; goto __kctl_end; }

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[alsa-devel] [patch] move dereference after null check
by Dan Carpenter 18 Nov '09

18 Nov '09

The 'jack' variable was dereference before the NULL check so this patch moves the dereference later. regards, dan carpenter Signed-off-by: Dan Carpenter <error27(a)gmail.com> --- orig/sound/soc/soc-jack.c 2009-11-14 11:04:04.000000000 +0200 +++ devel/sound/soc/soc-jack.c 2009-11-14 11:05:01.000000000 +0200 @@ -58,7 +58,7 @@ */ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) { - struct snd_soc_codec *codec = jack->card->codec; + struct snd_soc_codec *codec; struct snd_soc_jack_pin *pin; int enable; int oldstatus; @@ -67,6 +67,7 @@ WARN_ON_ONCE(!jack); return; } + codec = jack->card->codec; mutex_lock(&codec->mutex);

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[alsa-devel] [PATCH] opti-miro: use variables directly in the probe function
by Krzysztof Helt 18 Nov '09

18 Nov '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Use the fm_port and mpu_port variables directly in a probe function. This completely eliminates a need to copy the fm_port value to the snd_miro structure. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/opti9xx/miro.c | 33 +++++++++++++++++---------------- 1 files changed, 17 insertions(+), 16 deletions(-) diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index 02e30d7..b8170ad 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -118,8 +118,6 @@ struct snd_miro { int dma1; int dma2; - long fm_port; - long mpu_port; int mpu_irq; @@ -757,7 +755,6 @@ static int __devinit snd_miro_init(struct snd_miro *chip, chip->irq = -1; chip->dma1 = -1; chip->dma2 = -1; - chip->fm_port = -1; chip->mpu_port = -1; chip->mpu_irq = -1; @@ -1261,7 +1258,6 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) } miro->wss_base = port; - miro->fm_port = fm_port; miro->mpu_port = mpu_port; miro->irq = irq; miro->mpu_irq = mpu_irq; @@ -1276,11 +1272,12 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) } } - if (miro->mpu_port == SNDRV_AUTO_PORT) { - if ((miro->mpu_port = snd_legacy_find_free_ioport(possible_mpu_ports, 2)) < 0) { + if (mpu_port == SNDRV_AUTO_PORT) { + mpu_port = snd_legacy_find_free_ioport(possible_mpu_ports, 2); + if (mpu_port < 0) { snd_card_free(card); snd_printk(KERN_ERR "unable to find a free MPU401 port\n"); - return -EBUSY; + return -EBUSY } } if (miro->irq == SNDRV_AUTO_IRQ) { @@ -1380,20 +1377,24 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) card->shortname, miro->name, pcm->name, miro->wss_base + 4, miro->irq, miro->dma1, miro->dma2); - if (miro->mpu_port <= 0 || miro->mpu_port == SNDRV_AUTO_PORT) + if (mpu_port <= 0 || mpu_port == SNDRV_AUTO_PORT) rmidi = NULL; - else - if ((error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, - miro->mpu_port, 0, miro->mpu_irq, IRQF_DISABLED, - &rmidi))) - snd_printk(KERN_WARNING "no MPU-401 device at 0x%lx?\n", miro->mpu_port); + else { + error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, + mpu_port, 0, miro->mpu_irq, IRQF_DISABLED, + &rmidi); + if (error < 0) + snd_printk(KERN_WARNING "no MPU-401 device at 0x%lx?\n", + mpu_port); + } - if (miro->fm_port > 0 && miro->fm_port != SNDRV_AUTO_PORT) { + if (fm_port > 0 && fm_port != SNDRV_AUTO_PORT) { struct snd_opl3 *opl3 = NULL; struct snd_opl4 *opl4; - if (snd_opl4_create(card, miro->fm_port, miro->fm_port - 8, + if (snd_opl4_create(card, fm_port, fm_port - 8, 2, &opl3, &opl4) < 0) - snd_printk(KERN_WARNING "no OPL4 device at 0x%lx\n", miro->fm_port); + snd_printk(KERN_WARNING "no OPL4 device at 0x%lx\n", + fm_port); } if ((error = snd_set_aci_init_values(miro)) < 0) { -- 1.6.4 ---------------------------------------------------------------------- Internetowi nie placa! Otworz Konto Direct. Kliknij >> http://link.interia.pl/f2448

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[alsa-devel] [PATCH] cs4236: update control names
by Krzysztof Helt 18 Nov '09

18 Nov '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Update control names to be more closer to their meaning. Change the "Mono" name to the "Beep" as this line is usually used to forward the PC beeper signal to sound card's output. Update names for both cs423x and wss. Clean up cs4235 controls according to the cs4235 doc. Rename some of the cs4235 controls to be consistent with the cs4236's ones. Also, delete one misnamed cs4231 register define. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This is the second version of the patch. The Master Digital Playback is not renamed to the Master DAC Playback. include/sound/cs4231-regs.h | 1 - sound/isa/cs423x/cs4236_lib.c | 49 +++++++++++++++++----------------------- sound/isa/wss/wss_lib.c | 8 +++--- 3 files changed, 25 insertions(+), 33 deletions(-) diff --git a/include/sound/cs4231-regs.h b/include/sound/cs4231-regs.h index 9264753..66d28c2 100644 --- a/include/sound/cs4231-regs.h +++ b/include/sound/cs4231-regs.h @@ -70,7 +70,6 @@ #define AD1845_PWR_DOWN 0x1b /* power down control */ #define CS4235_LEFT_MASTER 0x1b /* left master output control */ #define CS4231_REC_FORMAT 0x1c /* clock and data format - record - bits 7-0 MCE */ -#define CS4231_PLY_VAR_FREQ 0x1d /* playback variable frequency */ #define AD1845_CLOCK 0x1d /* crystal clock select and total power down */ #define CS4235_RIGHT_MASTER 0x1d /* right master output control */ #define CS4231_REC_UPR_CNT 0x1e /* record upper count */ diff --git a/sound/isa/cs423x/cs4236_lib.c b/sound/isa/cs423x/cs4236_lib.c index 1b1ad1c..4c4024a 100644 --- a/sound/isa/cs423x/cs4236_lib.c +++ b/sound/isa/cs423x/cs4236_lib.c @@ -777,7 +777,7 @@ CS4236_DOUBLE("Mic Playback Switch", 0, CS4236_DOUBLE("Mic Capture Switch", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 7, 7, 1, 1), CS4236_DOUBLE("Mic Volume", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 0, 0, 31, 1), -CS4236_DOUBLE("Mic Playback Boost", 0, +CS4236_DOUBLE("Mic Playback Boost (+20dB)", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 5, 5, 1, 0), WSS_DOUBLE("Line Playback Switch", 0, @@ -798,10 +798,10 @@ WSS_DOUBLE("CD Capture Switch", 0, CS4236_DOUBLE1("Mono Output Playback Switch", 0, CS4231_MONO_CTRL, CS4236_RIGHT_MIX_CTRL, 6, 7, 1, 1), -CS4236_DOUBLE1("Mono Playback Switch", 0, +CS4236_DOUBLE1("Beep Playback Switch", 0, CS4231_MONO_CTRL, CS4236_LEFT_MIX_CTRL, 7, 7, 1, 1), -WSS_SINGLE("Mono Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), -WSS_SINGLE("Mono Playback Bypass", 0, CS4231_MONO_CTRL, 5, 1, 0), +WSS_SINGLE("Beep Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), +WSS_SINGLE("Beep Bypass Playback Switch", 0, CS4231_MONO_CTRL, 5, 1, 0), WSS_DOUBLE("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), @@ -815,31 +815,27 @@ CS4236_DOUBLE1("Digital Loopback Playback Volume", 0, static struct snd_kcontrol_new snd_cs4235_controls[] = { -WSS_DOUBLE("Master Switch", 0, +WSS_DOUBLE("Master Playback Switch", 0, CS4235_LEFT_MASTER, CS4235_RIGHT_MASTER, 7, 7, 1, 1), -WSS_DOUBLE("Master Volume", 0, +WSS_DOUBLE("Master Playback Volume", 0, CS4235_LEFT_MASTER, CS4235_RIGHT_MASTER, 0, 0, 31, 1), CS4235_OUTPUT_ACCU("Playback Volume", 0), -CS4236_DOUBLE("Master Digital Playback Switch", 0, - CS4236_LEFT_MASTER, CS4236_RIGHT_MASTER, 7, 7, 1, 1), -CS4236_DOUBLE("Master Digital Capture Switch", 0, - CS4236_DAC_MUTE, CS4236_DAC_MUTE, 7, 6, 1, 1), -CS4236_MASTER_DIGITAL("Master Digital Volume", 0), - -WSS_DOUBLE("Master Digital Playback Switch", 1, +WSS_DOUBLE("Synth Playback Switch", 1, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), -WSS_DOUBLE("Master Digital Capture Switch", 1, +WSS_DOUBLE("Synth Capture Switch", 1, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 6, 6, 1, 1), -WSS_DOUBLE("Master Digital Volume", 1, +WSS_DOUBLE("Synth Volume", 1, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1), CS4236_DOUBLE("Capture Volume", 0, CS4236_LEFT_MIX_CTRL, CS4236_RIGHT_MIX_CTRL, 5, 5, 3, 1), -WSS_DOUBLE("PCM Switch", 0, +WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), +WSS_DOUBLE("PCM Capture Switch", 0, + CS4236_DAC_MUTE, CS4236_DAC_MUTE, 7, 6, 1, 1), WSS_DOUBLE("PCM Volume", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1), @@ -855,28 +851,25 @@ CS4236_DOUBLE("Mic Capture Switch", 0, CS4236_DOUBLE("Mic Playback Switch", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 6, 6, 1, 1), CS4236_SINGLE("Mic Volume", 0, CS4236_LEFT_MIC, 0, 31, 1), -CS4236_SINGLE("Mic Playback Boost", 0, CS4236_LEFT_MIC, 5, 1, 0), +CS4236_SINGLE("Mic Boost (+20dB)", 0, CS4236_LEFT_MIC, 5, 1, 0), -WSS_DOUBLE("Aux Playback Switch", 0, +WSS_DOUBLE("Line Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Aux Capture Switch", 0, +WSS_DOUBLE("Line Capture Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 6, 6, 1, 1), -WSS_DOUBLE("Aux Volume", 0, +WSS_DOUBLE("Line Volume", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1), -WSS_DOUBLE("Aux Playback Switch", 1, +WSS_DOUBLE("CD Playback Switch", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Aux Capture Switch", 1, +WSS_DOUBLE("CD Capture Switch", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 6, 6, 1, 1), -WSS_DOUBLE("Aux Volume", 1, +WSS_DOUBLE("CD Volume", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), -CS4236_DOUBLE1("Master Mono Switch", 0, - CS4231_MONO_CTRL, CS4236_RIGHT_MIX_CTRL, 6, 7, 1, 1), - -CS4236_DOUBLE1("Mono Switch", 0, +CS4236_DOUBLE1("Beep Playback Switch", 0, CS4231_MONO_CTRL, CS4236_LEFT_MIX_CTRL, 7, 7, 1, 1), -WSS_SINGLE("Mono Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), +WSS_SINGLE("Beep Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), WSS_DOUBLE("Analog Loopback Switch", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 7, 7, 1, 0), diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 705db09..5b9d6c1 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -2224,7 +2224,7 @@ WSS_DOUBLE_TLV("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, .get = snd_wss_get_mux, .put = snd_wss_put_mux, }, -WSS_DOUBLE("Mic Boost", 0, +WSS_DOUBLE("Mic Boost (+20dB)", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), WSS_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0), @@ -2235,14 +2235,14 @@ WSS_DOUBLE("Line Playback Switch", 0, WSS_DOUBLE_TLV("Line Playback Volume", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1, db_scale_5bit_12db_max), -WSS_SINGLE("Mono Playback Switch", 0, +WSS_SINGLE("Beep Playback Switch", 0, CS4231_MONO_CTRL, 7, 1, 1), -WSS_SINGLE_TLV("Mono Playback Volume", 0, +WSS_SINGLE_TLV("Beep Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1, db_scale_4bit), WSS_SINGLE("Mono Output Playback Switch", 0, CS4231_MONO_CTRL, 6, 1, 1), -WSS_SINGLE("Mono Output Playback Bypass", 0, +WSS_SINGLE("Beep Bypass Playback Switch", 0, CS4231_MONO_CTRL, 5, 1, 0), }; -- 1.6.4 ---------------------------------------------------------------------- Kawalerka za 79 000 zl! 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