From bc56eff1279d2f33a6afe74a673360ae1cd0d838 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 13 Apr 2006 10:16:08 +0200 Subject: [ALSA] add another Phase 26 quirk Add a quirk entry for the TerraTec Phase 26 with yet another product ID. Signed-off-by: Clemens Ladisch diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 0992a09..9351846 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1531,6 +1531,15 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + USB_DEVICE_VENDOR_SPEC(0x0ccd, 0x0014), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "TerraTec", + .product_name = "PHASE 26", + .ifnum = 3, + .type = QUIRK_MIDI_STANDARD_INTERFACE + } +}, +{ USB_DEVICE(0x0ccd, 0x0035), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Miditech", -- cgit v0.10.2 From 5c59e09d7e51f5781439aa6f1963076568fd1f4f Mon Sep 17 00:00:00 2001 From: Steven Finney Date: Thu, 13 Apr 2006 12:49:31 +0200 Subject: [ALSA] Handle the error correctly in SNDCTL_DSP_SETFMT ioctl Handle the error returned from snd_pcm_oss_get_formats() correctly in SNDCTL_DSP_SETFMT ioctl handler of PCM OSS emulation. Signed-off-by: Steven Finney Signed-off-by: Takashi Iwai diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index c5978d6..a7567b8 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1242,6 +1242,8 @@ static int snd_pcm_oss_set_format(struct snd_pcm_oss_file *pcm_oss_file, int for if (format != AFMT_QUERY) { formats = snd_pcm_oss_get_formats(pcm_oss_file); + if (formats < 0) + return formats; if (!(formats & format)) format = AFMT_U8; for (idx = 1; idx >= 0; --idx) { -- cgit v0.10.2 From a182ee9876c7826d0b8f7789cb5c38c5bfbec441 Mon Sep 17 00:00:00 2001 From: Rene Herman Date: Thu, 13 Apr 2006 12:57:11 +0200 Subject: [ALSA] continue on IS_ERR from platform device registration I previously only concerned myself with sound/isa. When I now checked for more platform_device_register_simple() usages in ALSA I found a couple more drivers that needed the same patches as already submitted for all the ISA drivers. This first one is the continue-on-iserr patch for sound/drivers. This gets them all. Signed-off-by: Rene Herman Signed-off-by: Takashi Iwai diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index e35fd57..ae0df54 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -675,10 +675,8 @@ static int __init alsa_card_dummy_init(void) continue; device = platform_device_register_simple(SND_DUMMY_DRIVER, i, NULL, 0); - if (IS_ERR(device)) { - err = PTR_ERR(device); - goto errout; - } + if (IS_ERR(device)) + continue; devices[i] = device; cards++; } @@ -686,14 +684,10 @@ static int __init alsa_card_dummy_init(void) #ifdef MODULE printk(KERN_ERR "Dummy soundcard not found or device busy\n"); #endif - err = -ENODEV; - goto errout; + snd_dummy_unregister_all(); + return -ENODEV; } return 0; - - errout: - snd_dummy_unregister_all(); - return err; } static void __exit alsa_card_dummy_exit(void) diff --git a/sound/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c index 9ea3059..da7ef26 100644 --- a/sound/drivers/mpu401/mpu401.c +++ b/sound/drivers/mpu401/mpu401.c @@ -251,10 +251,8 @@ static int __init alsa_card_mpu401_init(void) #endif device = platform_device_register_simple(SND_MPU401_DRIVER, i, NULL, 0); - if (IS_ERR(device)) { - err = PTR_ERR(device); - goto errout; - } + if (IS_ERR(device)) + continue; platform_devices[i] = device; snd_mpu401_devices++; } @@ -266,14 +264,10 @@ static int __init alsa_card_mpu401_init(void) #ifdef MODULE printk(KERN_ERR "MPU-401 device not found or device busy\n"); #endif - err = -ENODEV; - goto errout; + snd_mpu401_unregister_all(); + return -ENODEV; } return 0; - - errout: - snd_mpu401_unregister_all(); - return err; } static void __exit alsa_card_mpu401_exit(void) diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c index 1a7fbef..c01b4c5 100644 --- a/sound/drivers/serial-u16550.c +++ b/sound/drivers/serial-u16550.c @@ -996,10 +996,8 @@ static int __init alsa_card_serial_init(void) continue; device = platform_device_register_simple(SND_SERIAL_DRIVER, i, NULL, 0); - if (IS_ERR(device)) { - err = PTR_ERR(device); - goto errout; - } + if (IS_ERR(device)) + continue; devices[i] = device; cards++; } @@ -1007,14 +1005,10 @@ static int __init alsa_card_serial_init(void) #ifdef MODULE printk(KERN_ERR "serial midi soundcard not found or device busy\n"); #endif - err = -ENODEV; - goto errout; + snd_serial_unregister_all(); + return -ENODEV; } return 0; - - errout: - snd_serial_unregister_all(); - return err; } static void __exit alsa_card_serial_exit(void) diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c index a3ee306..26eb249 100644 --- a/sound/drivers/virmidi.c +++ b/sound/drivers/virmidi.c @@ -169,10 +169,8 @@ static int __init alsa_card_virmidi_init(void) continue; device = platform_device_register_simple(SND_VIRMIDI_DRIVER, i, NULL, 0); - if (IS_ERR(device)) { - err = PTR_ERR(device); - goto errout; - } + if (IS_ERR(device)) + continue; devices[i] = device; cards++; } @@ -180,14 +178,10 @@ static int __init alsa_card_virmidi_init(void) #ifdef MODULE printk(KERN_ERR "Card-VirMIDI soundcard not found or device busy\n"); #endif - err = -ENODEV; - goto errout; + snd_virmidi_unregister_all(); + return -ENODEV; } return 0; - - errout: - snd_virmidi_unregister_all(); - return err; } static void __exit alsa_card_virmidi_exit(void) -- cgit v0.10.2 From 8970ccda1ae3c3b4ddd5ce366ca2cd88356d664e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Apr 2006 12:50:40 +0200 Subject: [ALSA] hda-codec - Use model 'hp' for all HP laptops with AD1981HD Use model 'hp' for all HP laptops with AD1981HD codec. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index bcfca15..fedac28 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1330,12 +1330,8 @@ enum { AD1981_BASIC, AD1981_HP }; static struct hda_board_config ad1981_cfg_tbl[] = { { .modelname = "hp", .config = AD1981_HP }, - { .pci_subvendor = 0x103c, .pci_subdevice = 0x30aa, - .config = AD1981_HP }, /* HP nx6320 */ - { .pci_subvendor = 0x103c, .pci_subdevice = 0x309f, - .config = AD1981_HP }, /* HP nx9420 AngelFire */ - { .pci_subvendor = 0x103c, .pci_subdevice = 0x30a2, - .config = AD1981_HP }, /* HP nx9420 AngelFire */ + /* All HP models */ + { .pci_subvendor = 0x103c, .config = AD1981_HP }, { .modelname = "basic", .config = AD1981_BASIC }, {} }; -- cgit v0.10.2 From 1d606f1ae5b9d83d8749f21bc04842596104bf55 Mon Sep 17 00:00:00 2001 From: Adrian Bunk Date: Tue, 18 Apr 2006 13:37:08 +0200 Subject: [ALSA] sound/pci/: remove duplicate #include's There's no reason for #include'ing linux/dma-mapping.h more than once. Signed-off-by: Adrian Bunk Signed-off-by: Takashi Iwai diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index c6c8333..2aa5a7f 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -39,7 +39,6 @@ #include #include #include -#include #include #include diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 3e332f3..2208dbd 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -36,7 +36,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index e3ad17f..dd465a1 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -104,7 +104,6 @@ #include #include #include -#include #include #include diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 32f8415..cc20a37 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -56,7 +56,6 @@ #include #include #include -#include #include #include diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 9c90d90..92a84aa 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -41,7 +41,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index b5a0950..4fad0e8 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -28,7 +28,6 @@ #include #include #include -#include #include #include diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index 35875c8..f679779 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -30,7 +30,6 @@ #include #include #include -#include #include #include -- cgit v0.10.2 From 531213a93f0e75b934471bf5567babad4da1ff70 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Apr 2006 13:46:08 +0200 Subject: [ALSA] hda-codec - Add entry for Epox EP-5LDA+ GLi Added the SSID entry for Epox EP-5LDA+ GLi with ALC880 codec. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 66bbdb6..f0e9a9c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2148,6 +2148,7 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .pci_subvendor = 0x1025, .pci_subdevice = 0x0087, .config = ALC880_6ST_DIG }, { .pci_subvendor = 0x1297, .pci_subdevice = 0xc790, .config = ALC880_6ST_DIG }, /* Shuttle ST20G5 */ { .pci_subvendor = 0x1509, .pci_subdevice = 0x925d, .config = ALC880_6ST_DIG }, /* FIC P4M-915GD1 */ + { .pci_subvendor = 0x1695, .pci_subdevice = 0x4012, .config = ALC880_5ST_DIG }, /* Epox EP-5LDA+ GLi */ { .modelname = "asus", .config = ALC880_ASUS }, { .pci_subvendor = 0x1043, .pci_subdevice = 0x1964, .config = ALC880_ASUS_DIG }, -- cgit v0.10.2 From c128b82cf4095bb64aec435cf58d67fb78272f2f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Apr 2006 15:01:23 +0200 Subject: [ALSA] Fix double free in error path of miro driver Fixed the double free in error path of miro driver. Signed-off-by: Takashi Iwai diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index 83d64bc..e6bfcf7 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -1179,20 +1179,17 @@ static int __init snd_card_miro_aci_detect(struct snd_card *card, struct snd_mir /* force ACI into a known state */ for (i = 0; i < 3; i++) if (aci_cmd(miro, ACI_ERROR_OP, -1, -1) < 0) { - snd_card_free(card); 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_card_free(card); snd_printk(KERN_ERR "can't read aci id on 0x%lx.\n", miro->aci_port); return -ENXIO; } if ((miro->aci_version=aci_cmd(miro, ACI_READ_VERSION, -1, -1)) < 0) { - snd_card_free(card); snd_printk(KERN_ERR "can't read aci version on 0x%lx.\n", miro->aci_port); return -ENXIO; -- cgit v0.10.2 From 5732e7a2cece461252bfcf2653bb09ab88ba36c5 Mon Sep 17 00:00:00 2001 From: Charis Kouzinopoulos Date: Tue, 18 Apr 2006 15:42:29 +0200 Subject: [ALSA] Fix typos and add information about Jack support to Audiophile-Usb.txt Signed-off-by: Charis Kouzinopoulos Signed-off-by: Thibault Le Meur Signed-off-by: Takashi Iwai diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt index 4692c8e..b535c2a 100644 --- a/Documentation/sound/alsa/Audiophile-Usb.txt +++ b/Documentation/sound/alsa/Audiophile-Usb.txt @@ -1,4 +1,4 @@ - Guide to using M-Audio Audiophile USB with ALSA and Jack v1.2 + Guide to using M-Audio Audiophile USB with ALSA and Jack v1.3 ======================================================== Thibault Le Meur @@ -22,16 +22,16 @@ The device has 4 audio interfaces, and 2 MIDI ports: * Midi In (Mi) * Midi Out (Mo) -The internal DAC/ADC has the following caracteristics: +The internal DAC/ADC has the following characteristics: * sample depth of 16 or 24 bits * sample rate from 8kHz to 96kHz -* Two ports can't use different sample depths at the same time.Moreover, the +* Two ports can't use different sample depths at the same time. Moreover, the Audiophile USB documentation gives the following Warning: "Please exit any audio application running before switching between bit depths" Due to the USB 1.1 bandwidth limitation, a limited number of interfaces can be activated at the same time depending on the audio mode selected: - * 16-bit/48kHz ==> 4 channels in/ 4 channels out + * 16-bit/48kHz ==> 4 channels in/4 channels out - Ai+Ao+Di+Do * 24-bit/48kHz ==> 4 channels in/2 channels out, or 2 channels in/4 channels out @@ -41,8 +41,8 @@ activated at the same time depending on the audio mode selected: Important facts about the Digital interface: -------------------------------------------- - * The Do port additionnaly supports surround-encoded AC-3 and DTS passthrough, -though I haven't tested it under linux + * The Do port additionally supports surround-encoded AC-3 and DTS passthrough, +though I haven't tested it under Linux - Note that in this setup only the Do interface can be enabled * Apart from recording an audio digital stream, enabling the Di port is a way to synchronize the device to an external sample clock @@ -60,24 +60,23 @@ synchronization error (for instance sound played at an odd sample rate) The Audiophile USB MIDI ports will be automatically supported once the following modules have been loaded: * snd-usb-audio - * snd-seq * snd-seq-midi -No additionnal setting is required. +No additional setting is required. 2.2 - Audio ports ----------------- Audio functions of the Audiophile USB device are handled by the snd-usb-audio module. This module can work in a default mode (without any device-specific -parameter), or in an advanced mode with the device-specific parameter called +parameter), or in an "advanced" mode with the device-specific parameter called "device_setup". 2.2.1 - Default Alsa driver mode -The default behaviour of the snd-usb-audio driver is to parse the device +The default behavior of the snd-usb-audio driver is to parse the device capabilities at startup and enable all functions inside the device (including -all ports at any sample rates and any sample depths supported). This approach +all ports at any supported sample rates and sample depths). This approach has the advantage to let the driver easily switch from sample rates/depths automatically according to the need of the application claiming the device. @@ -114,9 +113,9 @@ gain). For people having this problem, the snd-usb-audio module has a new module parameter called "device_setup". -2.2.2.1 - Initializing the working mode of the Audiohile USB +2.2.2.1 - Initializing the working mode of the Audiophile USB -As far as the Audiohile USB device is concerned, this value let the user +As far as the Audiophile USB device is concerned, this value let the user specify: * the sample depth * the sample rate @@ -174,20 +173,20 @@ The parameter can be given: IMPORTANT NOTE WHEN SWITCHING CONFIGURATION: ------------------------------------------- - * You may need to _first_ intialize the module with the correct device_setup + * You may need to _first_ initialize the module with the correct device_setup parameter and _only_after_ turn on the Audiophile USB device * This is especially true when switching the sample depth: - - first trun off the device - - de-register the snd-usb-audio module - - change the device_setup parameter (by either manually reprobing the module - or changing modprobe.conf) + - first turn off the device + - de-register the snd-usb-audio module (modprobe -r) + - change the device_setup parameter by changing the device_setup + option in /etc/modprobe.conf - turn on the device 2.2.2.3 - Audiophile USB's device_setup structure If you want to understand the device_setup magic numbers for the Audiophile USB, you need some very basic understanding of binary computation. However, -this is not required to use the parameter and you may skip thi section. +this is not required to use the parameter and you may skip this section. The device_setup is one byte long and its structure is the following: @@ -231,11 +230,11 @@ Caution: 2.2.3 - USB implementation details for this device -You may safely skip this section if you're not interrested in driver +You may safely skip this section if you're not interested in driver development. -This section describes some internals aspect of the device and summarize the -data I got by usb-snooping the windows and linux drivers. +This section describes some internal aspects of the device and summarize the +data I got by usb-snooping the windows and Linux drivers. The M-Audio Audiophile USB has 7 USB Interfaces: a "USB interface": @@ -277,9 +276,9 @@ Here is a short description of the AltSettings capabilities: - 16-bit depth, 8-48kHz sample mode - Synch playback (Do), audio format type III IEC1937_AC-3 -In order to ensure a correct intialization of the device, the driver +In order to ensure a correct initialization of the device, the driver _must_know_ how the device will be used: - * if DTS is choosen, only Interface 2 with AltSet nb.6 must be + * if DTS is chosen, only Interface 2 with AltSet nb.6 must be registered * if 96KHz only AltSets nb.1 of each interface must be selected * if samples are using 24bits/48KHz then AltSet 2 must me used if @@ -290,7 +289,7 @@ _must_know_ how the device will be used: is not connected When device_setup is given as a parameter to the snd-usb-audio module, the -parse_audio_enpoint function uses a quirk called +parse_audio_endpoints function uses a quirk called "audiophile_skip_setting_quirk" in order to prevent AltSettings not corresponding to device_setup from being registered in the driver. @@ -317,9 +316,8 @@ However you may see the following warning message: using the "default" ALSA device. This is less efficient than it could be. Consider using a hardware device instead rather than using the plug layer." - 3.2 - Patching alsa to use direct pcm device -------------------------------------------- +-------------------------------------------- A patch for Jack by Andreas Steinmetz adds support for Big Endian devices. However it has not been included in the CVS tree. @@ -331,3 +329,32 @@ After having applied the patch you can run jackd with the following command line: % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1 +3.2 - Getting 2 input and/or output interfaces in Jack +------------------------------------------------------ + +As you can see, starting the Jack server this way will only enable 1 stereo +input (Di or Ai) and 1 stereo output (Ao or Do). + +This is due to the following restrictions: +* Jack can only open one capture device and one playback device at a time +* The Audiophile USB is seen as 2 (or three) Alsa devices: hw:1,0, hw:1,1 + (and optionally hw:1,2) +If you want to get Ai+Di and/or Ao+Do support with Jack, you would need to +combine the Alsa devices into one logical "complex" device. + +If you want to give it a try, I recommend reading the information from +this page: http://www.sound-man.co.uk/linuxaudio/ice1712multi.html +It is related to another device (ice1712) but can be adapted to suit +the Audiophile USB. + +Enabling multiple Audiophile USB interfaces for Jackd will certainly require: +* patching Jack with the previously mentioned "Big Endian" patch +* patching Jackd with the MMAP_COMPLEX patch (see the ice1712 page) +* patching the alsa-lib/src/pcm/pcm_multi.c file (see the ice1712 page) +* define a multi device (combination of hw:1,0 and hw:1,1) in your .asoundrc + file +* start jackd with this device + +I had no success in testing this for now, but this may be due to my OS +configuration. If you have any success with this kind of setup, please +drop me an email. -- cgit v0.10.2 From 8263c65fbee1347b2ab1d8c9380946808d09f579 Mon Sep 17 00:00:00 2001 From: Bastiaan Jacques Date: Tue, 18 Apr 2006 17:04:04 +0200 Subject: [ALSA] via82xx: add support for VIA VT8251 (AC'97) Add support for VIA VT8251 AC'97. Includes a workaround which ensures sound won't stop playing after one second of playback. Signed-off-by: Bastiaan Jacques Signed-off-by: Takashi Iwai diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 0f171dd..1b740dd 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -123,6 +123,7 @@ module_param(enable, bool, 0444); #define VIA_REV_8233A 0x40 /* 1 rec, 1 multi-pb, spdf */ #define VIA_REV_8235 0x50 /* 2 rec, 4 pb, 1 multi-pb, spdif */ #define VIA_REV_8237 0x60 +#define VIA_REV_8251 0x70 /* * Direct registers @@ -863,8 +864,15 @@ static snd_pcm_uframes_t snd_via8233_pcm_pointer(struct snd_pcm_substream *subst status = inb(VIADEV_REG(viadev, OFFSET_STATUS)); if (!(status & VIA_REG_STAT_ACTIVE)) { - res = 0; - goto unlock; + /* An apparent bug in the 8251 is worked around by sending + * a REG_CTRL_START. */ + if (chip->revision == VIA_REV_8251) + snd_via82xx_pcm_trigger(substream, + SNDRV_PCM_TRIGGER_START); + else { + res = 0; + goto unlock; + } } if (count & 0xffffff) { idx = count >> 24; @@ -2313,6 +2321,7 @@ static struct via823x_info via823x_cards[] __devinitdata = { { VIA_REV_8233A, "VIA 8233A", TYPE_VIA8233A }, { VIA_REV_8235, "VIA 8235", TYPE_VIA8233 }, { VIA_REV_8237, "VIA 8237", TYPE_VIA8233 }, + { VIA_REV_8251, "VIA 8251", TYPE_VIA8233 }, }; /* @@ -2342,6 +2351,7 @@ static int __devinit check_dxs_list(struct pci_dev *pci) { .subvendor = 0x1043, .subdevice = 0x810d, .action = VIA_DXS_SRC }, /* ASUS */ { .subvendor = 0x1043, .subdevice = 0x812a, .action = VIA_DXS_SRC }, /* ASUS A8V Deluxe */ { .subvendor = 0x1043, .subdevice = 0x8174, .action = VIA_DXS_SRC }, /* ASUS */ + { .subvendor = 0x1043, .subdevice = 0x81b9, .action = VIA_DXS_SRC }, /* ASUS A8V-MX */ { .subvendor = 0x1071, .subdevice = 0x8375, .action = VIA_DXS_NO_VRA }, /* Vobis/Yakumo/Mitac notebook */ { .subvendor = 0x1071, .subdevice = 0x8399, .action = VIA_DXS_NO_VRA }, /* Umax AB 595T (VIA K8N800A - VT8237) */ { .subvendor = 0x10cf, .subdevice = 0x118e, .action = VIA_DXS_ENABLE }, /* FSC laptop */ -- cgit v0.10.2 From 1a183131fe284e68194e66cc4ff49d5876501eb0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Apr 2006 21:23:47 +0200 Subject: [ALSA] intel8x0 - Disable ALI5455 SPDIF-input Disable the SPDIF-input on ALI5455, which causes Oops. Signed-off-by: Takashi Iwai diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index ebbf2cf..035d084 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -1293,6 +1293,7 @@ static int snd_intel8x0_ali_ac97spdifout_close(struct snd_pcm_substream *substre return 0; } +#if 0 // NYI static int snd_intel8x0_ali_spdifin_open(struct snd_pcm_substream *substream) { struct intel8x0 *chip = snd_pcm_substream_chip(substream); @@ -1308,7 +1309,6 @@ static int snd_intel8x0_ali_spdifin_close(struct snd_pcm_substream *substream) return 0; } -#if 0 // NYI static int snd_intel8x0_ali_spdifout_open(struct snd_pcm_substream *substream) { struct intel8x0 *chip = snd_pcm_substream_chip(substream); @@ -1435,6 +1435,7 @@ static struct snd_pcm_ops snd_intel8x0_ali_ac97spdifout_ops = { .pointer = snd_intel8x0_pcm_pointer, }; +#if 0 // NYI static struct snd_pcm_ops snd_intel8x0_ali_spdifin_ops = { .open = snd_intel8x0_ali_spdifin_open, .close = snd_intel8x0_ali_spdifin_close, @@ -1446,7 +1447,6 @@ static struct snd_pcm_ops snd_intel8x0_ali_spdifin_ops = { .pointer = snd_intel8x0_pcm_pointer, }; -#if 0 // NYI static struct snd_pcm_ops snd_intel8x0_ali_spdifout_ops = { .open = snd_intel8x0_ali_spdifout_open, .close = snd_intel8x0_ali_spdifout_close, @@ -1582,7 +1582,7 @@ static struct ich_pcm_table ali_pcms[] __devinitdata = { { .suffix = "IEC958", .playback_ops = &snd_intel8x0_ali_ac97spdifout_ops, - .capture_ops = &snd_intel8x0_ali_spdifin_ops, + /* .capture_ops = &snd_intel8x0_ali_spdifin_ops, */ .prealloc_size = 64 * 1024, .prealloc_max_size = 128 * 1024, .ac97_idx = ALID_AC97SPDIFOUT, -- cgit v0.10.2 From c6cc0e3b0c087d350bdc5912ecdfb17e796ae266 Mon Sep 17 00:00:00 2001 From: Bastiaan Jacques Date: Thu, 20 Apr 2006 12:27:09 +0200 Subject: [ALSA] via82xx: tweak VT8251 workaround Move the workaround for the VT8251 up a bit, and check for STAT_EOL rather than STAT_ACTIVE. This resolves issues some people were having with certain ALSA clients (and allows the STAT_ACTIVE check to do what it was intended to do). This change was suggested by Andrew Daviel. Signed-off-by: Bastiaan Jacques Signed-off-by: Takashi Iwai diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 1b740dd..f7a22aa 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -863,16 +863,14 @@ static snd_pcm_uframes_t snd_via8233_pcm_pointer(struct snd_pcm_substream *subst if (!status) status = inb(VIADEV_REG(viadev, OFFSET_STATUS)); + /* An apparent bug in the 8251 is worked around by sending a + * REG_CTRL_START. */ + if (chip->revision == VIA_REV_8251 && (status & VIA_REG_STAT_EOL)) + snd_via82xx_pcm_trigger(substream, SNDRV_PCM_TRIGGER_START); + if (!(status & VIA_REG_STAT_ACTIVE)) { - /* An apparent bug in the 8251 is worked around by sending - * a REG_CTRL_START. */ - if (chip->revision == VIA_REV_8251) - snd_via82xx_pcm_trigger(substream, - SNDRV_PCM_TRIGGER_START); - else { - res = 0; - goto unlock; - } + res = 0; + goto unlock; } if (count & 0xffffff) { idx = count >> 24; -- cgit v0.10.2 From 711ee39bf3e2a69005d64f388441a6f883495f83 Mon Sep 17 00:00:00 2001 From: Henrik Kretzschmar Date: Thu, 20 Apr 2006 12:37:00 +0200 Subject: [ALSA] pcxhr - Fix a compiler warning on 64bit architectures The patch fixes a conpile warning on 64bit architectures, caused by different sizes of size_t . Since size_t is unsigned I permited myself to cange the format, too. Signed-off-by: Henrik Kretzschmar Signed-off-by: Takashi Iwai diff --git a/sound/pci/pcxhr/pcxhr_hwdep.c b/sound/pci/pcxhr/pcxhr_hwdep.c index 03517c1..369c19f 100644 --- a/sound/pci/pcxhr/pcxhr_hwdep.c +++ b/sound/pci/pcxhr/pcxhr_hwdep.c @@ -385,8 +385,8 @@ static int pcxhr_hwdep_dsp_load(struct snd_hwdep *hw, fw.size = dsp->length; fw.data = vmalloc(fw.size); if (! fw.data) { - snd_printk(KERN_ERR "pcxhr: cannot allocate dsp image (%d bytes)\n", - fw.size); + snd_printk(KERN_ERR "pcxhr: cannot allocate dsp image (%lu bytes)\n", + (unsigned long)fw.size); return -ENOMEM; } if (copy_from_user(fw.data, dsp->image, dsp->length)) { -- cgit v0.10.2 From 61a7454a229d3516492fc3ff3adddf9f5ac0d396 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 20 Apr 2006 16:42:34 +0200 Subject: [ALSA] hda-codec - Add model entry for ASUS M9 laptop Add a model entry to support ASUS M9 laptop with AD1986A codec. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index fedac28..522ffa7 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -799,6 +799,8 @@ static struct hda_board_config ad1986a_cfg_tbl[] = { { .modelname = "laptop-eapd", .config = AD1986A_LAPTOP_EAPD }, { .pci_subvendor = 0x144d, .pci_subdevice = 0xc024, .config = AD1986A_LAPTOP_EAPD }, /* Samsung R65-T2300 Charis */ + { .pci_subvendor = 0x1043, .pci_subdevice = 0x1153, + .config = AD1986A_LAPTOP_EAPD }, /* ASUS M9 */ { .pci_subvendor = 0x1043, .pci_subdevice = 0x1213, .config = AD1986A_LAPTOP_EAPD }, /* ASUS A6J */ { .pci_subvendor = 0x1043, .pci_subdevice = 0x11f7, -- cgit v0.10.2 From 71b2ccc3a2fd6c27e3cd9b4239670005978e94ce Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 21 Apr 2006 16:09:31 +0200 Subject: [ALSA] hda-codec - Add codec id for AD1988B codec chip Add codec id for AD1988B codec chip. The functionality is identical with AD1988(A) chip. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 522ffa7..f336cff 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2621,5 +2621,6 @@ struct hda_codec_preset snd_hda_preset_analog[] = { { .id = 0x11d41983, .name = "AD1983", .patch = patch_ad1983 }, { .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a }, { .id = 0x11d41988, .name = "AD1988", .patch = patch_ad1988 }, + { .id = 0x11d4198b, .name = "AD1988B", .patch = patch_ad1988 }, {} /* terminator */ }; -- cgit v0.10.2 From 396c9b928d5c24775846a161a8191dcc1ea4971f Mon Sep 17 00:00:00 2001 From: Henrik Kretzschmar Date: Mon, 24 Apr 2006 15:59:04 +0200 Subject: [ALSA] add __devinitdata to all pci_device_id Signed-off-by: Henrik Kretzschmar Signed-off-by: Takashi Iwai diff --git a/drivers/media/video/cx88/cx88-alsa.c b/drivers/media/video/cx88/cx88-alsa.c index f9d87b8..320b3d9 100644 --- a/drivers/media/video/cx88/cx88-alsa.c +++ b/drivers/media/video/cx88/cx88-alsa.c @@ -616,7 +616,7 @@ static struct snd_kcontrol_new snd_cx88_capture_volume = { * Only boards with eeprom and byte 1 at eeprom=1 have it */ -static struct pci_device_id cx88_audio_pci_tbl[] = { +static struct pci_device_id cx88_audio_pci_tbl[] __devinitdata = { {0x14f1,0x8801,PCI_ANY_ID,PCI_ANY_ID,0,0,0}, {0x14f1,0x8811,PCI_ANY_ID,PCI_ANY_ID,0,0,0}, {0, } diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index 2aa5a7f..eece1c7 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -1051,7 +1051,7 @@ snd_ad1889_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_device_id snd_ad1889_ids[] = { +static struct pci_device_id snd_ad1889_ids[] __devinitdata = { { PCI_DEVICE(PCI_VENDOR_ID_ANALOG_DEVICES, PCI_DEVICE_ID_AD1889JS) }, { 0, }, }; diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index fc92b68..e2dbc21 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -279,7 +279,7 @@ struct snd_ali { #endif }; -static struct pci_device_id snd_ali_ids[] = { +static struct pci_device_id snd_ali_ids[] __devinitdata = { {PCI_DEVICE(PCI_VENDOR_ID_AL, PCI_DEVICE_ID_AL_M5451), 0, 0, 0}, {0, } }; diff --git a/sound/pci/als300.c b/sound/pci/als300.c index 91899f8..901b08a 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -146,7 +146,7 @@ struct snd_als300_substream_data { int block_counter_register; }; -static struct pci_device_id snd_als300_ids[] = { +static struct pci_device_id snd_als300_ids[] __devinitdata = { { 0x4005, 0x0300, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_ALS300 }, { 0x4005, 0x0308, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_ALS300_PLUS }, { 0, } diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index 100d812..60423b1 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -116,7 +116,7 @@ struct snd_card_als4000 { #endif }; -static struct pci_device_id snd_als4000_ids[] = { +static struct pci_device_id snd_als4000_ids[] __devinitdata = { { 0x4005, 0x4000, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* ALS4000 */ { 0, } }; diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 12e6188..d0f759d 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -284,7 +284,7 @@ struct atiixp { /* */ -static struct pci_device_id snd_atiixp_ids[] = { +static struct pci_device_id snd_atiixp_ids[] __devinitdata = { { 0x1002, 0x4341, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB200 */ { 0x1002, 0x4361, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB300 */ { 0x1002, 0x4370, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB400 */ diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index 1d37660..12a34c3 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -262,7 +262,7 @@ struct atiixp_modem { /* */ -static struct pci_device_id snd_atiixp_ids[] = { +static struct pci_device_id snd_atiixp_ids[] __devinitdata = { { 0x1002, 0x434d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB200 */ { 0x1002, 0x4378, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB400 */ { 0, } diff --git a/sound/pci/au88x0/au8810.c b/sound/pci/au88x0/au8810.c index fce22c7..bd33529 100644 --- a/sound/pci/au88x0/au8810.c +++ b/sound/pci/au88x0/au8810.c @@ -1,6 +1,6 @@ #include "au8810.h" #include "au88x0.h" -static struct pci_device_id snd_vortex_ids[] = { +static struct pci_device_id snd_vortex_ids[] __devinitdata = { {PCI_VENDOR_ID_AUREAL, PCI_DEVICE_ID_AUREAL_ADVANTAGE, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 1,}, {0,} diff --git a/sound/pci/au88x0/au8820.c b/sound/pci/au88x0/au8820.c index d1fbcce..7e3fd83 100644 --- a/sound/pci/au88x0/au8820.c +++ b/sound/pci/au88x0/au8820.c @@ -1,6 +1,6 @@ #include "au8820.h" #include "au88x0.h" -static struct pci_device_id snd_vortex_ids[] = { +static struct pci_device_id snd_vortex_ids[] __devinitdata = { {PCI_VENDOR_ID_AUREAL, PCI_DEVICE_ID_AUREAL_VORTEX_1, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0,}, {0,} diff --git a/sound/pci/au88x0/au8830.c b/sound/pci/au88x0/au8830.c index d4f2717..b840f66 100644 --- a/sound/pci/au88x0/au8830.c +++ b/sound/pci/au88x0/au8830.c @@ -1,6 +1,6 @@ #include "au8830.h" #include "au88x0.h" -static struct pci_device_id snd_vortex_ids[] = { +static struct pci_device_id snd_vortex_ids[] __devinitdata = { {PCI_VENDOR_ID_AUREAL, PCI_DEVICE_ID_AUREAL_VORTEX_2, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0,}, {0,} diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 680077e..52a3645 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -216,7 +216,7 @@ struct snd_azf3328 { int irq; }; -static const struct pci_device_id snd_azf3328_ids[] = { +static const struct pci_device_id snd_azf3328_ids[] __devinitdata = { { 0x122D, 0x50DC, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* PCI168/3328 */ { 0x122D, 0x80DA, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* 3328 */ { 0, } diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 7b44a8d..9ee07d4 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -774,7 +774,7 @@ static int __devinit snd_bt87x_create(struct snd_card *card, .driver_data = rate } /* driver_data is the default digital_rate value for that device */ -static struct pci_device_id snd_bt87x_ids[] = { +static struct pci_device_id snd_bt87x_ids[] __devinitdata = { /* Hauppauge WinTV series */ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0x13eb, 32000), /* Hauppauge WinTV series */ @@ -911,7 +911,7 @@ static void __devexit snd_bt87x_remove(struct pci_dev *pci) /* default entries for all Bt87x cards - it's not exported */ /* driver_data is set to 0 to call detection */ -static struct pci_device_id snd_bt87x_default_ids[] = { +static struct pci_device_id snd_bt87x_default_ids[] __devinitdata = { BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, PCI_ANY_ID, PCI_ANY_ID, 0), BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_879, PCI_ANY_ID, PCI_ANY_ID, 0), { } diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 9477838..fd8bfeb 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -1561,7 +1561,7 @@ static void __devexit snd_ca0106_remove(struct pci_dev *pci) } // PCI IDs -static struct pci_device_id snd_ca0106_ids[] = { +static struct pci_device_id snd_ca0106_ids[] __devinitdata = { { 0x1102, 0x0007, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* Audigy LS or Live 24bit */ { 0, } }; diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 2ecbddb..e5ce2da 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -2609,7 +2609,7 @@ static inline void snd_cmipci_proc_init(struct cmipci *cm) {} #endif -static struct pci_device_id snd_cmipci_ids[] = { +static struct pci_device_id snd_cmipci_ids[] __devinitdata = { {PCI_VENDOR_ID_CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8338A, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, {PCI_VENDOR_ID_CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8338B, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, {PCI_VENDOR_ID_CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8738, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index ac4e73f..b3c94d8 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -494,7 +494,7 @@ struct cs4281 { static irqreturn_t snd_cs4281_interrupt(int irq, void *dev_id, struct pt_regs *regs); -static struct pci_device_id snd_cs4281_ids[] = { +static struct pci_device_id snd_cs4281_ids[] __devinitdata = { { 0x1013, 0x6005, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* CS4281 */ { 0, } }; diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c index c590602..848d772 100644 --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -65,7 +65,7 @@ MODULE_PARM_DESC(thinkpad, "Force to enable Thinkpad's CLKRUN control."); module_param_array(mmap_valid, bool, NULL, 0444); MODULE_PARM_DESC(mmap_valid, "Support OSS mmap."); -static struct pci_device_id snd_cs46xx_ids[] = { +static struct pci_device_id snd_cs46xx_ids[] __devinitdata = { { 0x1013, 0x6001, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* CS4280 */ { 0x1013, 0x6003, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* CS4612 */ { 0x1013, 0x6004, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* CS4615 */ diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index 9fc7f38..2c1213a 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -45,7 +45,7 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; -static struct pci_device_id snd_cs5535audio_ids[] = { +static struct pci_device_id snd_cs5535audio_ids[] __devinitdata = { { PCI_VENDOR_ID_NS, PCI_DEVICE_ID_NS_CS5535_AUDIO, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, { PCI_VENDOR_ID_AMD, PCI_DEVICE_ID_AMD_CS5536_AUDIO, diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 2dfa932..42b11ba 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -77,7 +77,7 @@ MODULE_PARM_DESC(subsystem, "Force card subsystem model."); /* * Class 0401: 1102:0008 (rev 00) Subsystem: 1102:1001 -> Audigy2 Value Model:SB0400 */ -static struct pci_device_id snd_emu10k1_ids[] = { +static struct pci_device_id snd_emu10k1_ids[] __devinitdata = { { 0x1102, 0x0002, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* EMU10K1 */ { 0x1102, 0x0004, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 1 }, /* Audigy */ { 0x1102, 0x0008, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 1 }, /* Audigy 2 Value SB0400 */ diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 2208dbd..d51290c 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1595,7 +1595,7 @@ static void __devexit snd_emu10k1x_remove(struct pci_dev *pci) } // PCI IDs -static struct pci_device_id snd_emu10k1x_ids[] = { +static struct pci_device_id snd_emu10k1x_ids[] __devinitdata = { { 0x1102, 0x0006, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* Dell OEM version (EMU10K1) */ { 0, } }; diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index a5533c8..ca9e34e 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -446,7 +446,7 @@ struct ensoniq { static irqreturn_t snd_audiopci_interrupt(int irq, void *dev_id, struct pt_regs *regs); -static struct pci_device_id snd_audiopci_ids[] = { +static struct pci_device_id snd_audiopci_ids[] __devinitdata = { #ifdef CHIP1370 { 0x1274, 0x5000, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* ES1370 */ #endif diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index 4d62fe4..6f9094c 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -242,7 +242,7 @@ struct es1938 { static irqreturn_t snd_es1938_interrupt(int irq, void *dev_id, struct pt_regs *regs); -static struct pci_device_id snd_es1938_ids[] = { +static struct pci_device_id snd_es1938_ids[] __devinitdata = { { 0x125d, 0x1969, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* Solo-1 */ { 0, } }; diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index dd465a1..5ff4175 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -592,7 +592,7 @@ struct es1968 { static irqreturn_t snd_es1968_interrupt(int irq, void *dev_id, struct pt_regs *regs); -static struct pci_device_id snd_es1968_ids[] = { +static struct pci_device_id snd_es1968_ids[] __devinitdata = { /* Maestro 1 */ { 0x1285, 0x0100, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, TYPE_MAESTRO }, /* Maestro 2 */ diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 6ab4aef..d72fc28 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -199,7 +199,7 @@ struct fm801 { #endif }; -static struct pci_device_id snd_fm801_ids[] = { +static struct pci_device_id snd_fm801_ids[] __devinitdata = { { 0x1319, 0x0801, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0, }, /* FM801 */ { 0x5213, 0x0510, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0, }, /* Gallant Odyssey Sound 4 */ { 0, } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0ad60ae..e821d65 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1614,7 +1614,7 @@ static void __devexit azx_remove(struct pci_dev *pci) } /* PCI IDs */ -static struct pci_device_id azx_ids[] = { +static struct pci_device_id azx_ids[] __devinitdata = { { 0x8086, 0x2668, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH6 */ { 0x8086, 0x27d8, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH7 */ { 0x8086, 0x269a, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ESB2 */ diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index cc20a37..c56793b 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -107,7 +107,7 @@ module_param_array(dxr_enable, int, NULL, 0444); MODULE_PARM_DESC(dxr_enable, "Enable DXR support for Terratec DMX6FIRE."); -static struct pci_device_id snd_ice1712_ids[] = { +static struct pci_device_id snd_ice1712_ids[] __devinitdata = { { PCI_VENDOR_ID_ICE, PCI_DEVICE_ID_ICE_1712, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* ICE1712 */ { 0, } }; diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index fce616c..b1c007e 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -86,7 +86,7 @@ MODULE_PARM_DESC(model, "Use the given board model."); /* Both VT1720 and VT1724 have the same PCI IDs */ -static struct pci_device_id snd_vt1724_ids[] = { +static struct pci_device_id snd_vt1724_ids[] __devinitdata = { { PCI_VENDOR_ID_ICE, PCI_DEVICE_ID_VT1724, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, { 0, } }; diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 035d084..0df7602 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -413,7 +413,7 @@ struct intel8x0 { u32 int_sta_mask; /* interrupt status mask */ }; -static struct pci_device_id snd_intel8x0_ids[] = { +static struct pci_device_id snd_intel8x0_ids[] __devinitdata = { { 0x8086, 0x2415, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 82801AA */ { 0x8086, 0x2425, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 82901AB */ { 0x8086, 0x2445, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 82801BA */ diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 47e26aa..720635f 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -224,7 +224,7 @@ struct intel8x0m { unsigned int pcm_pos_shift; }; -static struct pci_device_id snd_intel8x0m_ids[] = { +static struct pci_device_id snd_intel8x0m_ids[] __devinitdata = { { 0x8086, 0x2416, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 82801AA */ { 0x8086, 0x2426, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 82901AB */ { 0x8086, 0x2446, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 82801BA */ diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 4721c09..e39fad1 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -424,7 +424,7 @@ module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Korg 1212 soundcard."); MODULE_AUTHOR("Haroldo Gamal "); -static struct pci_device_id snd_korg1212_ids[] = { +static struct pci_device_id snd_korg1212_ids[] __devinitdata = { { .vendor = 0x10b5, .device = 0x906d, diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 92a84aa..1928e06 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -869,7 +869,7 @@ struct snd_m3 { /* * pci ids */ -static struct pci_device_id snd_m3_ids[] = { +static struct pci_device_id snd_m3_ids[] __devinitdata = { {PCI_VENDOR_ID_ESS, PCI_DEVICE_ID_ESS_ALLEGRO_1, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0}, {PCI_VENDOR_ID_ESS, PCI_DEVICE_ID_ESS_ALLEGRO, PCI_ANY_ID, PCI_ANY_ID, diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 4fad0e8..09cc078 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -61,7 +61,7 @@ MODULE_PARM_DESC(enable, "Enable Digigram " CARD_NAME " soundcard."); /* */ -static struct pci_device_id snd_mixart_ids[] = { +static struct pci_device_id snd_mixart_ids[] __devinitdata = { { 0x1057, 0x0003, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* MC8240 */ { 0, } }; diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index cc297ab..b92d660 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -263,7 +263,7 @@ struct nm256 { /* * PCI ids */ -static struct pci_device_id snd_nm256_ids[] = { +static struct pci_device_id snd_nm256_ids[] __devinitdata = { {PCI_VENDOR_ID_NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256AV_AUDIO, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, {PCI_VENDOR_ID_NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256ZX_AUDIO, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, {PCI_VENDOR_ID_NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256XL_PLUS_AUDIO, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index f679779..dafa223 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -73,7 +73,7 @@ enum { PCI_ID_LAST }; -static struct pci_device_id pcxhr_ids[] = { +static struct pci_device_id pcxhr_ids[] __devinitdata = { { 0x10b5, 0x9656, 0x1369, 0xb001, 0, 0, PCI_ID_VX882HR, }, /* VX882HR */ { 0x10b5, 0x9656, 0x1369, 0xb101, 0, 0, PCI_ID_PCX882HR, }, /* PCX882HR */ { 0x10b5, 0x9656, 0x1369, 0xb201, 0, 0, PCI_ID_VX881HR, }, /* VX881HR */ diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index f148ee4..d8cc985 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -506,7 +506,7 @@ static int riptide_reset(struct cmdif *cif, struct snd_riptide *chip); /* */ -static struct pci_device_id snd_riptide_ids[] = { +static struct pci_device_id snd_riptide_ids[] __devinitdata = { { .vendor = 0x127a,.device = 0x4310, .subvendor = PCI_ANY_ID,.subdevice = PCI_ANY_ID, @@ -527,7 +527,7 @@ static struct pci_device_id snd_riptide_ids[] = { }; #ifdef SUPPORT_JOYSTICK -static struct pci_device_id snd_riptide_joystick_ids[] = { +static struct pci_device_id snd_riptide_joystick_ids[] __devinitdata = { { .vendor = 0x127a,.device = 0x4312, .subvendor = PCI_ANY_ID,.subdevice = PCI_ANY_ID, diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index ab78544..55b1d48 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -227,7 +227,7 @@ struct rme32 { struct snd_kcontrol *spdif_ctl; }; -static struct pci_device_id snd_rme32_ids[] = { +static struct pci_device_id snd_rme32_ids[] __devinitdata = { {PCI_VENDOR_ID_XILINX_RME, PCI_DEVICE_ID_RME_DIGI32, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0,}, {PCI_VENDOR_ID_XILINX_RME, PCI_DEVICE_ID_RME_DIGI32_8, diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 6c2a9f4..3c1bc53 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -232,7 +232,7 @@ struct rme96 { struct snd_kcontrol *spdif_ctl; }; -static struct pci_device_id snd_rme96_ids[] = { +static struct pci_device_id snd_rme96_ids[] __devinitdata = { { PCI_VENDOR_ID_XILINX, PCI_DEVICE_ID_RME_DIGI96, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, { PCI_VENDOR_ID_XILINX, PCI_DEVICE_ID_RME_DIGI96_8, diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index ebf7a2b..61f82f0 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -568,7 +568,7 @@ static void snd_hammerfall_free_buffer(struct snd_dma_buffer *dmab, struct pci_d } -static struct pci_device_id snd_hdsp_ids[] = { +static struct pci_device_id snd_hdsp_ids[] __devinitdata = { { .vendor = PCI_VENDOR_ID_XILINX, .device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP, diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index b5538ef..722b9e6 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -426,7 +426,7 @@ static char channel_map_madi_qs[HDSPM_MAX_CHANNELS] = { }; -static struct pci_device_id snd_hdspm_ids[] = { +static struct pci_device_id snd_hdspm_ids[] __devinitdata = { { .vendor = PCI_VENDOR_ID_XILINX, .device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP_MADI, diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index a687eb6..75d6406 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -315,7 +315,7 @@ static void snd_hammerfall_free_buffer(struct snd_dma_buffer *dmab, struct pci_d } -static struct pci_device_id snd_rme9652_ids[] = { +static struct pci_device_id snd_rme9652_ids[] __devinitdata = { { .vendor = 0x10ee, .device = 0x3fc4, diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index 2d66a09..91f8bf3 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -243,7 +243,7 @@ struct sonicvibes { #endif }; -static struct pci_device_id snd_sonic_ids[] = { +static struct pci_device_id snd_sonic_ids[] __devinitdata = { { 0x5333, 0xca00, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, { 0, } }; diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c index b453804..9624a5f 100644 --- a/sound/pci/trident/trident.c +++ b/sound/pci/trident/trident.c @@ -63,7 +63,7 @@ MODULE_PARM_DESC(pcm_channels, "Number of hardware channels assigned for PCM."); module_param_array(wavetable_size, int, NULL, 0444); MODULE_PARM_DESC(wavetable_size, "Maximum memory size in kB for wavetable synth."); -static struct pci_device_id snd_trident_ids[] = { +static struct pci_device_id snd_trident_ids[] __devinitdata = { {PCI_DEVICE(PCI_VENDOR_ID_TRIDENT, PCI_DEVICE_ID_TRIDENT_4DWAVE_DX), PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0}, {PCI_DEVICE(PCI_VENDOR_ID_TRIDENT, PCI_DEVICE_ID_TRIDENT_4DWAVE_NX), diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index f7a22aa..111dada 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -396,7 +396,7 @@ struct via82xx { #endif }; -static struct pci_device_id snd_via82xx_ids[] = { +static struct pci_device_id snd_via82xx_ids[] __devinitdata = { /* 0x1106, 0x3058 */ { PCI_VENDOR_ID_VIA, PCI_DEVICE_ID_VIA_82C686_5, PCI_ANY_ID, PCI_ANY_ID, 0, 0, TYPE_CARD_VIA686, }, /* 686A */ /* 0x1106, 0x3059 */ diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 22ce4d3..ef97e50 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -261,7 +261,7 @@ struct via82xx_modem { struct snd_info_entry *proc_entry; }; -static struct pci_device_id snd_via82xx_modem_ids[] = { +static struct pci_device_id snd_via82xx_modem_ids[] __devinitdata = { { 0x1106, 0x3068, PCI_ANY_ID, PCI_ANY_ID, 0, 0, TYPE_CARD_VIA82XX_MODEM, }, { 0, } }; diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c index c816ddf..0f1ebb0 100644 --- a/sound/pci/vx222/vx222.c +++ b/sound/pci/vx222/vx222.c @@ -60,7 +60,7 @@ enum { VX_PCI_VX222_NEW }; -static struct pci_device_id snd_vx222_ids[] = { +static struct pci_device_id snd_vx222_ids[] __devinitdata = { { 0x10b5, 0x9050, 0x1369, PCI_ANY_ID, 0, 0, VX_PCI_VX222_OLD, }, /* PLX */ { 0x10b5, 0x9030, 0x1369, PCI_ANY_ID, 0, 0, VX_PCI_VX222_NEW, }, /* PLX */ { 0, } diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index db57ce9..65ebf5f 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -70,7 +70,7 @@ MODULE_PARM_DESC(rear_switch, "Enable shared rear/line-in switch"); module_param_array(rear_swap, bool, NULL, 0444); MODULE_PARM_DESC(rear_swap, "Swap rear channels (must be enabled for correct IEC958 (S/PDIF)) output"); -static struct pci_device_id snd_ymfpci_ids[] = { +static struct pci_device_id snd_ymfpci_ids[] __devinitdata = { { 0x1073, 0x0004, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* YMF724 */ { 0x1073, 0x000d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* YMF724F */ { 0x1073, 0x000a, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* YMF740 */ -- cgit v0.10.2 From a2bbbc0c3c9554ac70e96ec3effae60124f0f009 Mon Sep 17 00:00:00 2001 From: Kenrik Kretzschmar Date: Mon, 24 Apr 2006 15:59:38 +0200 Subject: [ALSA] adding __devinitdata to pci_device_id Refering to /Documentation/pci.txt the struct pci_device_id can be released after loading the module. Signed-off-by: Kenrik Kretzschmar Signed-off-by: Takashi Iwai diff --git a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl index 68eeebc..1faf763 100644 --- a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl +++ b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl @@ -1172,7 +1172,7 @@ } /* PCI IDs */ - static struct pci_device_id snd_mychip_ids[] = { + static struct pci_device_id snd_mychip_ids[] __devinitdata = { { PCI_VENDOR_ID_FOO, PCI_DEVICE_ID_BAR, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, .... @@ -1565,7 +1565,7 @@ Date: Mon, 24 Apr 2006 21:57:16 +0200 Subject: [ALSA] PCM core - introduce CONFIG_SND_PCM_XRUN_DEBUG This patch makes the XRUN (overrun/underrun) notification code optional. Signed-off-by: Jaroslav Kysela diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 8efc1b1..f3a07fb 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -171,3 +171,13 @@ config SND_DEBUG_DETECT help Say Y here to enable extra-verbose log messages printed when detecting devices. + +config SND_PCM_XRUN_DEBUG + bool "Enable PCM ring buffer overrun/underrun debugging" + default n + depends on SND_DEBUG + help + Say Y to enable the PCM ring buffer overrun/underrun debugging. + It is usually not required, but if you have trouble with + sound clicking when system is loaded, it may help to determine + the process or driver which causes the scheduling gaps. diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 122e10a..48007a5 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -436,7 +436,7 @@ static void snd_pcm_substream_proc_status_read(struct snd_info_entry *entry, snd_iprintf(buffer, "appl_ptr : %ld\n", runtime->control->appl_ptr); } -#ifdef CONFIG_SND_DEBUG +#ifdef CONFIG_SND_PCM_XRUN_DEBUG static void snd_pcm_xrun_debug_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { @@ -480,7 +480,7 @@ static int snd_pcm_stream_proc_init(struct snd_pcm_str *pstr) } pstr->proc_info_entry = entry; -#ifdef CONFIG_SND_DEBUG +#ifdef CONFIG_SND_PCM_XRUN_DEBUG if ((entry = snd_info_create_card_entry(pcm->card, "xrun_debug", pstr->proc_root)) != NULL) { entry->c.text.read_size = 64; @@ -501,7 +501,7 @@ static int snd_pcm_stream_proc_init(struct snd_pcm_str *pstr) static int snd_pcm_stream_proc_done(struct snd_pcm_str *pstr) { -#ifdef CONFIG_SND_DEBUG +#ifdef CONFIG_SND_PCM_XRUN_DEBUG if (pstr->proc_xrun_debug_entry) { snd_info_unregister(pstr->proc_xrun_debug_entry); pstr->proc_xrun_debug_entry = NULL; diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 230a940..eedc6cb 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -130,7 +130,7 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_ufram static void xrun(struct snd_pcm_substream *substream) { snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); -#ifdef CONFIG_SND_DEBUG +#ifdef CONFIG_SND_PCM_XRUN_DEBUG if (substream->pstr->xrun_debug) { snd_printd(KERN_DEBUG "XRUN: pcmC%dD%d%c\n", substream->pcm->card->number, @@ -204,7 +204,7 @@ static inline int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *subs delta = hw_ptr_interrupt - new_hw_ptr; if (delta > 0) { if ((snd_pcm_uframes_t)delta < runtime->buffer_size / 2) { -#ifdef CONFIG_SND_DEBUG +#ifdef CONFIG_SND_PCM_XRUN_DEBUG if (runtime->periods > 1 && substream->pstr->xrun_debug) { snd_printd(KERN_ERR "Unexpected hw_pointer value [1] (stream = %i, delta: -%ld, max jitter = %ld): wrong interrupt acknowledge?\n", substream->stream, (long) delta, runtime->buffer_size / 2); if (substream->pstr->xrun_debug > 1) @@ -249,7 +249,7 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) delta = old_hw_ptr - new_hw_ptr; if (delta > 0) { if ((snd_pcm_uframes_t)delta < runtime->buffer_size / 2) { -#ifdef CONFIG_SND_DEBUG +#ifdef CONFIG_SND_PCM_XRUN_DEBUG if (runtime->periods > 2 && substream->pstr->xrun_debug) { snd_printd(KERN_ERR "Unexpected hw_pointer value [2] (stream = %i, delta: -%ld, max jitter = %ld): wrong interrupt acknowledge?\n", substream->stream, (long) delta, runtime->buffer_size / 2); if (substream->pstr->xrun_debug > 1) -- cgit v0.10.2 From b7d90a356a43f4609bd6290fc2e1ca4ef79d4458 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 25 Apr 2006 12:56:04 +0200 Subject: [ALSA] Fix Oops at rmmod with CONFIG_SND_VERBOSE_PROCFS=n Fixed Oops at rmmod with CONFIG_SND_VERBOSE_PROCFS=n. Add ifdef to struct fields for optimization and better compile checks. Signed-off-by: Takashi Iwai diff --git a/include/sound/pcm.h b/include/sound/pcm.h index df70e75..3734258 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -374,12 +374,14 @@ struct snd_pcm_substream { /* -- OSS things -- */ struct snd_pcm_oss_substream oss; #endif +#ifdef CONFIG_SND_VERBOSE_PROCFS struct snd_info_entry *proc_root; struct snd_info_entry *proc_info_entry; struct snd_info_entry *proc_hw_params_entry; struct snd_info_entry *proc_sw_params_entry; struct snd_info_entry *proc_status_entry; struct snd_info_entry *proc_prealloc_entry; +#endif /* misc flags */ unsigned int no_mmap_ctrl: 1; unsigned int hw_opened: 1; @@ -400,12 +402,14 @@ struct snd_pcm_str { struct snd_pcm_oss_stream oss; #endif struct snd_pcm_file *files; +#ifdef CONFIG_SND_VERBOSE_PROCFS struct snd_info_entry *proc_root; struct snd_info_entry *proc_info_entry; -#ifdef CONFIG_SND_DEBUG +#ifdef CONFIG_SND_PCM_XRUN_DEBUG unsigned int xrun_debug; /* 0 = disabled, 1 = verbose, 2 = stacktrace */ struct snd_info_entry *proc_xrun_debug_entry; #endif +#endif }; struct snd_pcm { diff --git a/include/sound/pcm_oss.h b/include/sound/pcm_oss.h index 39df2ba..c854647 100644 --- a/include/sound/pcm_oss.h +++ b/include/sound/pcm_oss.h @@ -75,7 +75,9 @@ struct snd_pcm_oss_substream { struct snd_pcm_oss_stream { struct snd_pcm_oss_setup *setup_list; /* setup list */ struct mutex setup_mutex; +#ifdef CONFIG_SND_VERBOSE_PROCFS struct snd_info_entry *proc_entry; +#endif }; struct snd_pcm_oss { diff --git a/sound/core/Kconfig b/sound/core/Kconfig index f3a07fb..4262a1c 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -142,7 +142,7 @@ config SND_SUPPORT_OLD_API config SND_VERBOSE_PROCFS bool "Verbose procfs contents" - depends on SND + depends on SND && PROC_FS default y help Say Y here to include code for verbose procfs contents (provides @@ -175,7 +175,7 @@ config SND_DEBUG_DETECT config SND_PCM_XRUN_DEBUG bool "Enable PCM ring buffer overrun/underrun debugging" default n - depends on SND_DEBUG + depends on SND_DEBUG && SND_VERBOSE_PROCFS help Say Y to enable the PCM ring buffer overrun/underrun debugging. It is usually not required, but if you have trouble with diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index a7567b8..ac990bf 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2214,7 +2214,7 @@ static int snd_pcm_oss_mmap(struct file *file, struct vm_area_struct *area) return 0; } -#ifdef CONFIG_PROC_FS +#ifdef CONFIG_SND_VERBOSE_PROCFS /* * /proc interface */ @@ -2368,10 +2368,10 @@ static void snd_pcm_oss_proc_done(struct snd_pcm *pcm) } } } -#else /* !CONFIG_PROC_FS */ +#else /* !CONFIG_SND_VERBOSE_PROCFS */ #define snd_pcm_oss_proc_init(pcm) #define snd_pcm_oss_proc_done(pcm) -#endif /* CONFIG_PROC_FS */ +#endif /* CONFIG_SND_VERBOSE_PROCFS */ /* * ENTRY functions diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 48007a5..84b0003 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -142,7 +142,7 @@ static int snd_pcm_control_ioctl(struct snd_card *card, return -ENOIOCTLCMD; } -#if defined(CONFIG_PROC_FS) && defined(CONFIG_SND_VERBOSE_PROCFS) +#ifdef CONFIG_SND_VERBOSE_PROCFS #define STATE(v) [SNDRV_PCM_STATE_##v] = #v #define STREAM(v) [SNDRV_PCM_STREAM_##v] = #v @@ -599,12 +599,12 @@ static int snd_pcm_substream_proc_done(struct snd_pcm_substream *substream) } return 0; } -#else /* !CONFIG_PROC_FS */ +#else /* !CONFIG_SND_VERBOSE_PROCFS */ static inline int snd_pcm_stream_proc_init(struct snd_pcm_str *pstr) { return 0; } static inline int snd_pcm_stream_proc_done(struct snd_pcm_str *pstr) { return 0; } static inline int snd_pcm_substream_proc_init(struct snd_pcm_substream *substream) { return 0; } static inline int snd_pcm_substream_proc_done(struct snd_pcm_substream *substream) { return 0; } -#endif /* CONFIG_PROC_FS */ +#endif /* CONFIG_SND_VERBOSE_PROCFS */ /** * snd_pcm_new_stream - create a new PCM stream diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index a0119ae..428f8c1 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -100,8 +100,10 @@ static void snd_pcm_lib_preallocate_dma_free(struct snd_pcm_substream *substream int snd_pcm_lib_preallocate_free(struct snd_pcm_substream *substream) { snd_pcm_lib_preallocate_dma_free(substream); +#ifdef CONFIG_SND_VERBOSE_PROCFS snd_info_unregister(substream->proc_prealloc_entry); substream->proc_prealloc_entry = NULL; +#endif return 0; } @@ -124,7 +126,7 @@ int snd_pcm_lib_preallocate_free_for_all(struct snd_pcm *pcm) return 0; } -#ifdef CONFIG_PROC_FS +#ifdef CONFIG_SND_VERBOSE_PROCFS /* * read callback for prealloc proc file * @@ -203,9 +205,9 @@ static inline void preallocate_info_init(struct snd_pcm_substream *substream) substream->proc_prealloc_entry = entry; } -#else /* !CONFIG_PROC_FS */ +#else /* !CONFIG_SND_VERBOSE_PROCFS */ #define preallocate_info_init(s) -#endif +#endif /* CONFIG_SND_VERBOSE_PROCFS */ /* * pre-allocate the buffer and create a proc file for the substream -- cgit v0.10.2 From d773781cee3e3003c4a77589d327de802b26e101 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 25 Apr 2006 13:05:43 +0200 Subject: [ALSA] hda-codec - Fix capture from line-in on VAIO SZ/FE laptops Added the missing line-in capture on VAIO SZ/FE laptops with STAC 7661 codec. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7152607..8c440fb 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1212,8 +1212,8 @@ static hda_nid_t vaio_mux_nids[] = { 0x15 }; static struct hda_input_mux vaio_mux = { .num_items = 2, .items = { - /* { "HP", 0x0 }, - { "Unknown", 0x1 }, */ + /* { "HP", 0x0 }, */ + { "Line", 0x1 }, { "Mic", 0x2 }, { "PCM", 0x3 }, } -- cgit v0.10.2 From c93dd4451ef68a5494c376944bdff8d75a8625bc Mon Sep 17 00:00:00 2001 From: Erik Mouw Date: Tue, 25 Apr 2006 13:08:59 +0200 Subject: [ALSA] PCMCIA sound devices shouldn't depend on ISA The ALSA drivers for PCMCIA devices depend on ISA, but modern laptops can have PCMCIA support without ISA. This patch removes the dependency. Signed-off-by: Erik Mouw Signed-off-by: Takashi Iwai diff --git a/sound/pcmcia/Kconfig b/sound/pcmcia/Kconfig index 5d1b0b7..c9fa1a2 100644 --- a/sound/pcmcia/Kconfig +++ b/sound/pcmcia/Kconfig @@ -5,7 +5,7 @@ menu "PCMCIA devices" config SND_VXPOCKET tristate "Digigram VXpocket" - depends on SND && PCMCIA && ISA + depends on SND && PCMCIA select SND_VX_LIB help Say Y here to include support for Digigram VXpocket and @@ -16,7 +16,7 @@ config SND_VXPOCKET config SND_PDAUDIOCF tristate "Sound Core PDAudioCF" - depends on SND && PCMCIA && ISA + depends on SND && PCMCIA select SND_PCM help Say Y here to include support for Sound Core PDAudioCF -- cgit v0.10.2 From e0292bdd306a7e1ef7a681350cf0427688a2791d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 26 Apr 2006 17:44:18 +0200 Subject: [ALSA] hda-codec - Add model entry for ASUS Z62F Added a model entry 'laptop-eapd' for ASUS Z62F laptop with AD1986A codec. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index f336cff..40f000b 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -805,6 +805,8 @@ static struct hda_board_config ad1986a_cfg_tbl[] = { .config = AD1986A_LAPTOP_EAPD }, /* ASUS A6J */ { .pci_subvendor = 0x1043, .pci_subdevice = 0x11f7, .config = AD1986A_LAPTOP_EAPD }, /* ASUS U5A */ + { .pci_subvendor = 0x1043, .pci_subdevice = 0x1297, + .config = AD1986A_LAPTOP_EAPD }, /* ASUS Z62F */ { .pci_subvendor = 0x103c, .pci_subdevice = 0x30af, .config = AD1986A_LAPTOP_EAPD }, /* HP Compaq Presario B2800 */ {} -- cgit v0.10.2 From a769577b3716c757e354a681aab3524ac6b651be Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 27 Apr 2006 16:56:07 +0200 Subject: [ALSA] via82xx - Use DXS_SRC as default for VIA8235/8237/8251 chips Use DXS_SRC as the default value for dxs_support option for VIA8235/8237/8251 chips. These new chips should work well with SRC. For VIA8233/A/C, the old default DXS_48K is still used to be sure. Signed-off-by: Takashi Iwai diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 111dada..39daf62 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -2332,7 +2332,7 @@ struct dxs_whitelist { short action; /* new dxs_support value */ }; -static int __devinit check_dxs_list(struct pci_dev *pci) +static int __devinit check_dxs_list(struct pci_dev *pci, int revision) { static struct dxs_whitelist whitelist[] = { { .subvendor = 0x1005, .subdevice = 0x4710, .action = VIA_DXS_ENABLE }, /* Avance Logic Mobo */ @@ -2413,6 +2413,10 @@ static int __devinit check_dxs_list(struct pci_dev *pci) } } + /* for newer revision, default to DXS_SRC */ + if (revision >= VIA_REV_8235) + return VIA_DXS_SRC; + /* * not detected, try 48k rate only to be sure. */ @@ -2457,7 +2461,7 @@ static int __devinit snd_via82xx_probe(struct pci_dev *pci, } if (chip_type != TYPE_VIA8233A) { if (dxs_support == VIA_DXS_AUTO) - dxs_support = check_dxs_list(pci); + dxs_support = check_dxs_list(pci, revision); /* force to use VIA8233 or 8233A model according to * dxs_support module option */ -- cgit v0.10.2