diff options
370 files changed, 11183 insertions, 5968 deletions
@@ -1933,7 +1933,7 @@ M: seasons@makosteszta.sote.hu D: Original author of software suspend N: Jaroslav Kysela -E: perex@suse.cz +E: perex@perex.cz W: http://www.perex.cz D: Original Author and Maintainer for HP 10/100 Mbit Network Adapters D: ISA PnP diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 241e26c4ff92..4b48c2e82c3c 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -365,13 +365,14 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module snd-cmipci ----------------- - Module for C-Media CMI8338 and 8738 PCI sound cards. + Module for C-Media CMI8338/8738/8768/8770 PCI sound cards. - mpu_port - 0x300,0x310,0x320,0x330 = legacy port, - 1 = integrated PCI port, + mpu_port - port address of MIDI interface (8338 only): + 0x300,0x310,0x320,0x330 = legacy port, 0 = disable (default) - fm_port - 0x388 = legacy port, - 1 = integrated PCI port (default), + fm_port - port address of OPL-3 FM synthesizer (8x38 only): + 0x388 = legacy port, + 1 = integrated PCI port (default on 8738), 0 = disable soft_ac3 - Software-conversion of raw SPDIF packets (model 033 only) (default = 1) @@ -768,6 +769,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. single_cmd - Use single immediate commands to communicate with codecs (for debugging only) enable_msi - Enable Message Signaled Interrupt (MSI) (default = off) + power_save - Automatic power-saving timtout (in second, 0 = + disable) + power_save_controller - Reset HD-audio controller in power-saving mode + (default = on) This module supports one card and autoprobe. @@ -828,6 +833,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. ALC268 3stack 3-stack model + toshiba Toshiba A205 + acer Acer laptops auto auto-config reading BIOS (default) ALC662 @@ -842,7 +849,11 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 3stack-dig 3-jack with SPDIF I/O 6stack-dig 6-jack digital with SPDIF I/O arima Arima W820Di1 + targa Targa T8, MSI-1049 T8 + asus-a7j ASUS A7J + asus-a7m ASUS A7M macpro MacPro support + mbp3 Macbook Pro rev3 imac24 iMac 24'' with jack detection w2jc ASUS W2JC auto auto-config reading BIOS (default) @@ -854,6 +865,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 3stack-6ch-dig 3-jack 6-channel with SPDIF I/O 6stack-dig-demo 6-jack digital for Intel demo board acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc) + acer-aspire Acer Aspire 9810 medion Medion Laptops medion-md2 Medion MD2 targa-dig Targa/MSI @@ -862,6 +874,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. lenovo-101e Lenovo 101E lenovo-nb0763 Lenovo NB0763 lenovo-ms7195-dig Lenovo MS7195 + haier-w66 Haier W66 6stack-hp HP machines with 6stack (Nettle boards) 3stack-hp HP machines with 3stack (Lucknow, Samba boards) auto auto-config reading BIOS (default) @@ -885,6 +898,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 3stack-660-digout 3-jack with SPDIF OUT (for ALC660VD) lenovo Lenovo 3000 C200 dallas Dallas laptops + hp HP TX1000 auto auto-config reading BIOS (default) CMI9880 @@ -920,6 +934,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 3stack 3-stack, shared surrounds laptop 2-channel only (FSC V2060, Samsung M50) laptop-eapd 2-channel with EAPD (Samsung R65, ASUS A6J) + laptop-automute 2-channel with EAPD and HP-automute (Lenovo N100) ultra 2-channel with EAPD (Samsung Ultra tablet PC) AD1988 @@ -945,14 +960,30 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. can be adjusted. Appearing only when compiled with $CONFIG_SND_DEBUG=y - STAC9200/9205/9254 + STAC9200 ref Reference board + dell-d21 Dell (unknown) + dell-d22 Dell (unknown) + dell-d23 Dell (unknown) + dell-m21 Dell Inspiron 630m, Dell Inspiron 640m + dell-m22 Dell Latitude D620, Dell Latitude D820 + dell-m23 Dell XPS M1710, Dell Precision M90 + dell-m24 Dell Latitude 120L + dell-m25 Dell Inspiron E1505n + dell-m26 Dell Inspiron 1501 + dell-m27 Dell Inspiron E1705/9400 + gateway Gateway laptops with EAPD control + + STAC9205/9254 + ref Reference board + dell-m42 Dell (unknown) + dell-m43 Dell Precision + dell-m44 Dell Inspiron STAC9220/9221 ref Reference board 3stack D945 3stack 5stack D945 5stack + SPDIF - dell Dell XPS M1210 intel-mac-v1 Intel Mac Type 1 intel-mac-v2 Intel Mac Type 2 intel-mac-v3 Intel Mac Type 3 @@ -964,6 +995,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. macbook-pro Intel Mac Book Pro 2nd generation (eq. type 3) imac-intel Intel iMac (eq. type 2) imac-intel-20 Intel iMac (newer version) (eq. type 3) + dell-d81 Dell (unknown) + dell-d82 Dell (unknown) + dell-m81 Dell (unknown) + dell-m82 Dell XPS M1210 STAC9202/9250/9251 ref Reference board, base config @@ -975,6 +1010,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. ref Reference board 3stack D965 3stack 5stack D965 5stack + SPDIF + dell-3stack Dell Dimension E520 STAC9872 vaio Setup for VAIO FE550G/SZ110 @@ -989,6 +1025,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. subsystem ID (output of "lspci -nv") to ALSA BTS or alsa-devel ML (see the section "Links and Addresses"). + power_save and power_save_controller options are for power-saving + mode. See powersave.txt for details. + Note 2: If you get click noises on output, try the module option position_fix=1 or 2. position_fix=1 will use the SD_LPIB register value without FIFO size correction as the current @@ -1349,7 +1388,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. port - port number or -1 (disable) irq - IRQ number or -1 (disable) pnp - PnP detection - 0 = disable, 1 = enable (default) - uart_enter - Issue UART_ENTER command at open - bool, default = on This module supports multiple devices and PnP. @@ -1630,6 +1668,21 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. The power-management is supported. + Module snd-sc6000 + ----------------- + + Module for Gallant SC-6000 soundcard. + + port - Port # (0x220 or 0x240) + mss_port - MSS Port # (0x530 or 0xe80) + irq - IRQ # (5,7,9,10,11) + mpu_irq - MPU-401 IRQ # (5,7,9,10) ,0 - no MPU-401 irq + dma - DMA # (1,3,0) + + This module supports multiple cards. + + This card is also known as Audio Excel DSP 16 or Zoltrix AV302. + Module snd-sgalaxy ------------------ @@ -1650,9 +1703,11 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module for ENSONIQ SoundScape PnP cards. port - Port # (PnP setup) + wss_port - WSS Port # (PnP setup) irq - IRQ # (PnP setup) mpu_irq - MPU-401 IRQ # (PnP setup) dma - DMA # (PnP setup) + dma2 - 2nd DMA # (PnP setup, -1 to disable) This module supports multiple cards. ISA PnP must be enabled. You need sscape_ctl tool in alsa-tools package for loading @@ -1697,8 +1752,52 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. dma2 - DMA2 # for CS4232 PCM interface. isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + The below are options for wavefront_synth features: + wf_raw - Assume that we need to boot the OS (default:no) + If yes, then during driver loading, the state of the board is + ignored, and we reset the board and load the firmware anyway. + fx_raw - Assume that the FX process needs help (default:yes) + If false, we'll leave the FX processor in whatever state it is + when the driver is loaded. The default is to download the + microprogram and associated coefficients to set it up for + "default" operation, whatever that means. + debug_default - Debug parameters for card initialization + wait_usecs - How long to wait without sleeping, usecs + (default:150) + This magic number seems to give pretty optimal throughput + based on my limited experimentation. + If you want to play around with it and find a better value, be + my guest. Remember, the idea is to get a number that causes us + to just busy wait for as many WaveFront commands as possible, + without coming up with a number so large that we hog the whole + CPU. + Specifically, with this number, out of about 134,000 status + waits, only about 250 result in a sleep. + sleep_interval - How long to sleep when waiting for reply + (default: 100) + sleep_tries - How many times to try sleeping during a wait + (default: 50) + ospath - Pathname to processed ICS2115 OS firmware + (default:wavefront.os) + The path name of the ISC2115 OS firmware. In the recent + version, it's handled via firmware loader framework, so it + must be installed in the proper path, typically, + /lib/firmware. + reset_time - How long to wait for a reset to take effect + (default:2) + ramcheck_time - How many seconds to wait for the RAM test + (default:20) + osrun_time - How many seconds to wait for the ICS2115 OS + (default:10) + This module supports multiple cards and ISA PnP. + Note: the firmware file "wavefront.os" was located in the earlier + version in /etc. Now it's loaded via firmware loader, and + must be in the proper firmware path, such as /lib/firmware. + Copy (or symlink) the file appropriately if you get an error + regarding firmware downloading after upgrading the kernel. + Module snd-sonicvibes --------------------- diff --git a/Documentation/sound/alsa/CMIPCI.txt b/Documentation/sound/alsa/CMIPCI.txt index 4b2b15387056..16935c8561f7 100644 --- a/Documentation/sound/alsa/CMIPCI.txt +++ b/Documentation/sound/alsa/CMIPCI.txt @@ -1,5 +1,5 @@ - Brief Notes on C-Media 8738/8338 Driver - ======================================= + Brief Notes on C-Media 8338/8738/8768/8770 Driver + ================================================= Takashi Iwai <tiwai@suse.de> @@ -209,10 +209,13 @@ In addition to the standard SB mixer, CM8x38 provides more functions. MIDI CONTROLLER --------------- -The MPU401-UART interface is disabled as default. You need to set -module option "mpu_port" with a valid I/O port address to enable the -MIDI support. The valid I/O ports are 0x300, 0x310, 0x320 and 0x330. -Choose the value which doesn't conflict with other cards. +With CMI8338 chips, the MPU401-UART interface is disabled as default. +You need to set the module option "mpu_port" to a valid I/O port address +to enable MIDI support. Valid I/O ports are 0x300, 0x310, 0x320 and +0x330. Choose a value that doesn't conflict with other cards. + +With CMI8738 and newer chips, the MIDI interface is enabled by default +and the driver automatically chooses a port address. There is _no_ hardware wavetable function on this chip (except for OPL3 synth below). @@ -230,6 +233,8 @@ Set "fm_port" module option for more cards. The output quality of FM OPL/3 is, however, very weird. I don't know why.. +CMI8768 and newer chips do not have the FM synth. + Joystick and Modem ------------------ diff --git a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl index 74d3a35b59bc..2c3fc3cb3b6b 100644 --- a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl +++ b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl @@ -18,8 +18,8 @@ </affiliation> </author> - <date>November 17, 2005</date> - <edition>0.3.6</edition> + <date>September 10, 2007</date> + <edition>0.3.7</edition> <abstract> <para> @@ -405,8 +405,9 @@ /* definition of the chip-specific record */ struct mychip { struct snd_card *card; - // rest of implementation will be in the section - // "PCI Resource Managements" + /* rest of implementation will be in the section + * "PCI Resource Managements" + */ }; /* chip-specific destructor @@ -414,7 +415,7 @@ */ static int snd_mychip_free(struct mychip *chip) { - .... // will be implemented later... + .... /* will be implemented later... */ } /* component-destructor @@ -440,8 +441,9 @@ *rchip = NULL; - // check PCI availability here - // (see "PCI Resource Managements") + /* check PCI availability here + * (see "PCI Resource Managements") + */ .... /* allocate a chip-specific data with zero filled */ @@ -451,12 +453,13 @@ chip->card = card; - // rest of initialization here; will be implemented - // later, see "PCI Resource Managements" + /* rest of initialization here; will be implemented + * later, see "PCI Resource Managements" + */ .... - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, - chip, &ops)) < 0) { + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { snd_mychip_free(chip); return err; } @@ -490,7 +493,8 @@ return -ENOMEM; /* (3) */ - if ((err = snd_mychip_create(card, pci, &chip)) < 0) { + err = snd_mychip_create(card, pci, &chip); + if (err < 0) { snd_card_free(card); return err; } @@ -502,10 +506,11 @@ card->shortname, chip->ioport, chip->irq); /* (5) */ - .... // implemented later + .... /* implemented later */ /* (6) */ - if ((err = snd_card_register(card)) < 0) { + err = snd_card_register(card); + if (err < 0) { snd_card_free(card); return err; } @@ -605,7 +610,8 @@ <![CDATA[ struct mychip *chip; .... - if ((err = snd_mychip_create(card, pci, &chip)) < 0) { + err = snd_mychip_create(card, pci, &chip); + if (err < 0) { snd_card_free(card); return err; } @@ -666,7 +672,8 @@ <informalexample> <programlisting> <![CDATA[ - if ((err = snd_card_register(card)) < 0) { + err = snd_card_register(card); + if (err < 0) { snd_card_free(card); return err; } @@ -1091,7 +1098,7 @@ static int snd_mychip_free(struct mychip *chip) { /* disable hardware here if any */ - .... // (not implemented in this document) + .... /* (not implemented in this document) */ /* release the irq */ if (chip->irq >= 0) @@ -1119,7 +1126,8 @@ *rchip = NULL; /* initialize the PCI entry */ - if ((err = pci_enable_device(pci)) < 0) + err = pci_enable_device(pci); + if (err < 0) return err; /* check PCI availability (28bit DMA) */ if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 || @@ -1141,7 +1149,8 @@ chip->irq = -1; /* (1) PCI resource allocation */ - if ((err = pci_request_regions(pci, "My Chip")) < 0) { + err = pci_request_regions(pci, "My Chip"); + if (err < 0) { kfree(chip); pci_disable_device(pci); return err; @@ -1156,10 +1165,10 @@ chip->irq = pci->irq; /* (2) initialization of the chip hardware */ - .... // (not implemented in this document) + .... /* (not implemented in this document) */ - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, - chip, &ops)) < 0) { + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { snd_mychip_free(chip); return err; } @@ -1233,7 +1242,8 @@ <informalexample> <programlisting> <![CDATA[ - if ((err = pci_enable_device(pci)) < 0) + err = pci_enable_device(pci); + if (err < 0) return err; if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 || pci_set_consistent_dma_mask(pci, DMA_28BIT_MASK) < 0) { @@ -1294,7 +1304,8 @@ <informalexample> <programlisting> <![CDATA[ - if ((err = pci_request_regions(pci, "My Chip")) < 0) { + err = pci_request_regions(pci, "My Chip"); + if (err < 0) { kfree(chip); pci_disable_device(pci); return err; @@ -1322,7 +1333,7 @@ <programlisting> <![CDATA[ if (request_irq(pci->irq, snd_mychip_interrupt, - IRQF_DISABLED|IRQF_SHARED, "My Chip", chip)) { + IRQF_SHARED, "My Chip", chip)) { printk(KERN_ERR "cannot grab irq %d\n", pci->irq); snd_mychip_free(chip); return -EBUSY; @@ -1773,7 +1784,8 @@ struct snd_pcm_runtime *runtime = substream->runtime; runtime->hw = snd_mychip_playback_hw; - // more hardware-initialization will be done here + /* more hardware-initialization will be done here */ + .... return 0; } @@ -1781,7 +1793,8 @@ static int snd_mychip_playback_close(struct snd_pcm_substream *substream) { struct mychip *chip = snd_pcm_substream_chip(substream); - // the hardware-specific codes will be here + /* the hardware-specific codes will be here */ + .... return 0; } @@ -1793,7 +1806,8 @@ struct snd_pcm_runtime *runtime = substream->runtime; runtime->hw = snd_mychip_capture_hw; - // more hardware-initialization will be done here + /* more hardware-initialization will be done here */ + .... return 0; } @@ -1801,7 +1815,8 @@ static int snd_mychip_capture_close(struct snd_pcm_substream *substream) { struct mychip *chip = snd_pcm_substream_chip(substream); - // the hardware-specific codes will be here + /* the hardware-specific codes will be here */ + .... return 0; } @@ -1844,10 +1859,12 @@ { switch (cmd) { case SNDRV_PCM_TRIGGER_START: - // do something to start the PCM engine + /* do something to start the PCM engine */ + .... break; case SNDRV_PCM_TRIGGER_STOP: - // do something to stop the PCM engine + /* do something to stop the PCM engine */ + .... break; default: return -EINVAL; @@ -1900,8 +1917,8 @@ struct snd_pcm *pcm; int err; - if ((err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1, - &pcm)) < 0) + err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1, &pcm); + if (err < 0) return err; pcm->private_data = chip; strcpy(pcm->name, "My Chip"); @@ -1939,8 +1956,8 @@ struct snd_pcm *pcm; int err; - if ((err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1, - &pcm)) < 0) + err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1, &pcm); + if (err < 0) return err; pcm->private_data = chip; strcpy(pcm->name, "My Chip"); @@ -2097,7 +2114,7 @@ struct mychip *chip = snd_pcm_chip(pcm); /* free your own data */ kfree(chip->my_private_pcm_data); - // do what you like else + /* do what you like else */ .... } @@ -2884,10 +2901,10 @@ struct _snd_pcm_runtime { <![CDATA[ switch (cmd) { case SNDRV_PCM_TRIGGER_START: - // do something to start the PCM engine + /* do something to start the PCM engine */ break; case SNDRV_PCM_TRIGGER_STOP: - // do something to stop the PCM engine + /* do something to stop the PCM engine */ break; default: return -EINVAL; @@ -3071,7 +3088,7 @@ struct _snd_pcm_runtime { spin_unlock(&chip->lock); snd_pcm_period_elapsed(chip->substream); spin_lock(&chip->lock); - // acknowledge the interrupt if necessary + /* acknowledge the interrupt if necessary */ } .... spin_unlock(&chip->lock); @@ -3134,7 +3151,7 @@ struct _snd_pcm_runtime { snd_pcm_period_elapsed(substream); spin_lock(&chip->lock); } - // acknowledge the interrupt if necessary + /* acknowledge the interrupt if necessary */ } .... spin_unlock(&chip->lock); @@ -3456,6 +3473,13 @@ struct _snd_pcm_runtime { </para> <para> + The <structfield>tlv</structfield> field can be used to provide + metadata about the control; see the + <link linkend="control-interface-tlv"> + <citetitle>Metadata</citetitle></link> subsection. + </para> + + <para> The other three are <link linkend="control-interface-callbacks"><citetitle> callback functions</citetitle></link>. @@ -3604,7 +3628,7 @@ struct _snd_pcm_runtime { <title>Example of info callback</title> <programlisting> <![CDATA[ - static int snd_myctl_info(struct snd_kcontrol *kcontrol, + static int snd_myctl_mono_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; @@ -3639,7 +3663,7 @@ struct _snd_pcm_runtime { <informalexample> <programlisting> <![CDATA[ - static int snd_myctl_info(struct snd_kcontrol *kcontrol, + static int snd_myctl_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[4] = { @@ -3658,6 +3682,16 @@ struct _snd_pcm_runtime { </programlisting> </informalexample> </para> + + <para> + Some common info callbacks are prepared for easy use: + <function>snd_ctl_boolean_mono_info()</function> and + <function>snd_ctl_boolean_stereo_info()</function>. + Obviously, the former is an info callback for a mono channel + boolean item, just like <function>snd_myctl_mono_info</function> + above, and the latter is for a stereo channel boolean item. + </para> + </section> <section id="control-interface-callbacks-get"> @@ -3794,7 +3828,8 @@ struct _snd_pcm_runtime { <informalexample> <programlisting> <![CDATA[ - if ((err = snd_ctl_add(card, snd_ctl_new1(&my_control, chip))) < 0) + err = snd_ctl_add(card, snd_ctl_new1(&my_control, chip)); + if (err < 0) return err; ]]> </programlisting> @@ -3843,6 +3878,56 @@ struct _snd_pcm_runtime { </para> </section> + <section id="control-interface-tlv"> + <title>Metadata</title> + <para> + To provide information about the dB values of a mixer control, use + on of the <constant>DECLARE_TLV_xxx</constant> macros from + <filename><sound/tlv.h></filename> to define a variable + containing this information, set the<structfield>tlv.p + </structfield> field to point to this variable, and include the + <constant>SNDRV_CTL_ELEM_ACCESS_TLV_READ</constant> flag in the + <structfield>access</structfield> field; like this: + <informalexample> + <programlisting> +<![CDATA[ + static DECLARE_TLV_DB_SCALE(db_scale_my_control, -4050, 150, 0); + + static struct snd_kcontrol_new my_control __devinitdata = { + ... + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + ... + .tlv.p = db_scale_my_control, + }; +]]> + </programlisting> + </informalexample> + </para> + + <para> + The <function>DECLARE_TLV_DB_SCALE</function> macro defines + information about a mixer control where each step in the control's + value changes the dB value by a constant dB amount. + The first parameter is the name of the variable to be defined. + The second parameter is the minimum value, in units of 0.01 dB. + The third parameter is the step size, in units of 0.01 dB. + Set the fourth parameter to 1 if the minimum value actually mutes + the control. + </para> + + <para> + The <function>DECLARE_TLV_DB_LINEAR</function> macro defines + information about a mixer control where the control's value affects + the output linearly. + The first parameter is the name of the variable to be defined. + The second parameter is the minimum value, in units of 0.01 dB. + The third parameter is the maximum value, in units of 0.01 dB. + If the minimum value mutes the control, set the second parameter to + <constant>TLV_DB_GAIN_MUTE</constant>. + </para> + </section> + </chapter> @@ -3880,7 +3965,7 @@ struct _snd_pcm_runtime { { struct mychip *chip = ac97->private_data; .... - // read a register value here from the codec + /* read a register value here from the codec */ return the_register_value; } @@ -3889,7 +3974,7 @@ struct _snd_pcm_runtime { { struct mychip *chip = ac97->private_data; .... - // write the given register value to the codec + /* write the given register value to the codec */ } static int snd_mychip_ac97(struct mychip *chip) @@ -3902,7 +3987,8 @@ struct _snd_pcm_runtime { .read = snd_mychip_ac97_read, }; - if ((err = snd_ac97_bus(chip->card, 0, &ops, NULL, &bus)) < 0) + err = snd_ac97_bus(chip->card, 0, &ops, NULL, &bus); + if (err < 0) return err; memset(&ac97, 0, sizeof(ac97)); ac97.private_data = chip; @@ -4447,10 +4533,10 @@ struct _snd_pcm_runtime { <informalexample> <programlisting> <![CDATA[ - struct list_head *list; struct snd_rawmidi_substream *substream; - list_for_each(list, &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams) { - substream = list_entry(list, struct snd_rawmidi_substream, list); + list_for_each_entry(substream, + &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams, + list { sprintf(substream->name, "My MIDI Port %d", substream->number + 1); } /* same for SNDRV_RAWMIDI_STREAM_INPUT */ diff --git a/Documentation/sound/alsa/OSS-Emulation.txt b/Documentation/sound/alsa/OSS-Emulation.txt index bfa0c9aacb4b..022aaeb0e9dd 100644 --- a/Documentation/sound/alsa/OSS-Emulation.txt +++ b/Documentation/sound/alsa/OSS-Emulation.txt @@ -303,10 +303,3 @@ ICE1712 supports only the unconventional format, interleaved the buffer as the conventional (mono or 2-channels, 8 or 16bit) format on OSS. -USB devices ------------ -Some USB devices support only 24bit format packed in 3bytes. This -format is not supported by OSS and no conversion is provided by kernel -OSS emulation. You can use the user-space OSS emulation via libaoss -instead. - diff --git a/Documentation/sound/alsa/hda_codec.txt b/Documentation/sound/alsa/hda_codec.txt index 4eaae2a45534..8e1b02526698 100644 --- a/Documentation/sound/alsa/hda_codec.txt +++ b/Documentation/sound/alsa/hda_codec.txt @@ -49,6 +49,9 @@ struct hda_bus_ops { unsigned int verb, unsigned int parm); unsigned int (*get_response)(struct hda_codec *codec); void (*private_free)(struct hda_bus *); +#ifdef CONFIG_SND_HDA_POWER_SAVE + void (*pm_notify)(struct hda_codec *codec); +#endif }; The command callback is called when the codec module needs to send a @@ -56,9 +59,16 @@ VERB to the controller. It's always a single command. The get_response callback is called when the codec requires the answer for the last command. These two callbacks are mandatory and have to be given. -The last, private_free callback, is optional. It's called in the +The third, private_free callback, is optional. It's called in the destructor to release any necessary data in the lowlevel driver. +The pm_notify callback is available only with +CONFIG_SND_HDA_POWER_SAVE kconfig. It's called when the codec needs +to power up or may power down. The controller should check the all +belonging codecs on the bus whether they are actually powered off +(check codec->power_on), and optionally the driver may power down the +contoller side, too. + The bus instance is created via snd_hda_bus_new(). You need to pass the card instance, the template, and the pointer to store the resultant bus instance. @@ -86,10 +96,8 @@ resultant codec instance (can be NULL if not needed). The codec is stored in a linked list of bus instance. You can follow the codec list like: - struct list_head *p; struct hda_codec *codec; - list_for_each(p, &bus->codec_list) { - codec = list_entry(p, struct hda_codec, list); + list_for_each_entry(codec, &bus->codec_list, list) { ... } @@ -100,10 +108,15 @@ initialization sequence is called when the controls are built later. Codec Access ============ -To access codec, use snd_codec_read() and snd_codec_write(). +To access codec, use snd_hda_codec_read() and snd_hda_codec_write(). snd_hda_param_read() is for reading parameters. For writing a sequence of verbs, use snd_hda_sequence_write(). +There are variants of cached read/write, snd_hda_codec_write_cache(), +snd_hda_sequence_write_cache(). These are used for recording the +register states for the power-mangement resume. When no PM is needed, +these are equivalent with non-cached version. + To retrieve the number of sub nodes connected to the given node, use snd_hda_get_sub_nodes(). The connection list can be obtained via snd_hda_get_connections() call. @@ -239,6 +252,10 @@ set the codec->patch_ops field. This is defined as below: int (*suspend)(struct hda_codec *codec, pm_message_t state); int (*resume)(struct hda_codec *codec); #endif + #ifdef CONFIG_SND_HDA_POWER_SAVE + int (*check_power_status)(struct hda_codec *codec, + hda_nid_t nid); + #endif }; The build_controls callback is called from snd_hda_build_controls(). @@ -251,6 +268,18 @@ The unsol_event callback is called when an unsolicited event is received. The suspend and resume callbacks are for power management. +They can be NULL if no special sequence is required. When the resume +callback is NULL, the driver calls the init callback and resumes the +registers from the cache. If other handling is needed, you'd need to +write your own resume callback. There, the amp values can be resumed +via + void snd_hda_codec_resume_amp(struct hda_codec *codec); +and the other codec registers via + void snd_hda_codec_resume_cache(struct hda_codec *codec); + +The check_power_status callback is called when the amp value of the +given widget NID is changed. The codec code can turn on/off the power +appropriately from this information. Each entry can be NULL if not necessary to be called. @@ -267,8 +296,7 @@ Digital I/O =========== Call snd_hda_create_spdif_out_ctls() from the patch to create controls -related with SPDIF out. In the patch resume callback, call -snd_hda_resume_spdif(). +related with SPDIF out. Helper Functions @@ -284,12 +312,7 @@ as a module parameter, and PCI subsystem IDs. If the matching entry is found, it returns the config field value. snd_hda_add_new_ctls() can be used to create and add control entries. -Pass the zero-terminated array of struct snd_kcontrol_new. The same array -can be passed to snd_hda_resume_ctls() for resume. -Note that this will call control->put callback of these entries. So, -put callback should check codec->in_resume and force to restore the -given value if it's non-zero even if the value is identical with the -cached value. +Pass the zero-terminated array of struct snd_kcontrol_new Macros HDA_CODEC_VOLUME(), HDA_CODEC_MUTE() and their variables can be used for the entry of struct snd_kcontrol_new. diff --git a/Documentation/sound/alsa/powersave.txt b/Documentation/sound/alsa/powersave.txt new file mode 100644 index 000000000000..9657e8099228 --- /dev/null +++ b/Documentation/sound/alsa/powersave.txt @@ -0,0 +1,41 @@ +Notes on Power-Saving Mode +========================== + +AC97 and HD-audio drivers have the automatic power-saving mode. +This feature is enabled via Kconfig CONFIG_SND_AC97_POWER_SAVE +and CONFIG_SND_HDA_POWER_SAVE options, respectively. + +With the automatic power-saving, the driver turns off the codec power +appropriately when no operation is required. When no applications use +the device and/or no analog loopback is set, the power disablement is +done fully or partially. It'll save a certain power consumption, thus +good for laptops (even for desktops). + +The time-out for automatic power-off can be specified via power_save +module option of snd-ac97-codec and snd-hda-intel modules. Specify +the time-out value in seconds. 0 means to disable the automatic +power-saving. The default value of timeout is given via +CONFIG_SND_AC97_POWER_SAVE_DEFAULT and +CONFIG_SND_HDA_POWER_SAVE_DEFAULT Kconfig options. Setting this to 1 +(the minimum value) isn't recommended because many applications try to +reopen the device frequently. 10 would be a good choice for normal +operations. + +The power_save option is exported as writable. This means you can +adjust the value via sysfs on the fly. For example, to turn on the +automatic power-save mode with 10 seconds, write to +/sys/modules/snd_ac97_codec/parameters/power_save (usually as root): + + # echo 10 > /sys/modules/snd_ac97_codec/parameters/power_save + + +Note that you might hear click noise/pop when changing the power +state. Also, it often takes certain time to wake up from the +power-down to the active state. These are often hardly to fix, so +don't report extra bug reports unless you have a fix patch ;-) + +For HD-audio interface, there is another module option, +power_save_controller. This enables/disables the power-save mode of +the controller side. Setting this on may reduce a bit more power +consumption, but might result in longer wake-up time and click noise. +Try to turn it off when you experience such a thing too often. diff --git a/MAINTAINERS b/MAINTAINERS index c7355e7f09ff..1315cca8fc5f 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -1769,7 +1769,7 @@ S: Maintained HP100: Driver for HP 10/100 Mbit/s Voice Grade Network Adapter Series P: Jaroslav Kysela -M: perex@suse.cz +M: perex@perex.cz S: Maintained HPET: High Precision Event Timers driver (hpet.c) @@ -2132,7 +2132,7 @@ S: Maintained ISAPNP P: Jaroslav Kysela -M: perex@suse.cz +M: perex@perex.cz S: Maintained ISDN SUBSYSTEM @@ -3523,7 +3523,7 @@ S: Maintained SOUND P: Jaroslav Kysela -M: perex@suse.cz +M: perex@perex.cz L: alsa-devel@alsa-project.org (subscribers-only) S: Maintained diff --git a/drivers/media/video/cx88/cx88-alsa.c b/drivers/media/video/cx88/cx88-alsa.c index 90c36c5705c3..141dadf7cf1b 100644 --- a/drivers/media/video/cx88/cx88-alsa.c +++ b/drivers/media/video/cx88/cx88-alsa.c @@ -7,7 +7,7 @@ * (c) 2005,2006 Ricardo Cerqueira <v4l@cerqueira.org> * (c) 2005 Mauro Carvalho Chehab <mchehab@infradead.org> * Based on a dummy cx88 module by Gerd Knorr <kraxel@bytesex.org> - * Based on dummy.c by Jaroslav Kysela <perex@suse.cz> + * Based on dummy.c by Jaroslav Kysela <perex@perex.cz> * * 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 diff --git a/drivers/net/hp100.c b/drivers/net/hp100.c index e4fde17e2841..49421d1cd3a5 100644 --- a/drivers/net/hp100.c +++ b/drivers/net/hp100.c @@ -8,7 +8,7 @@ ** Extended for new busmaster capable chipsets by ** Siegfried "Frieder" Loeffler (dg1sek) <floeff@mathematik.uni-stuttgart.de> ** -** Maintained by: Jaroslav Kysela <perex@suse.cz> +** Maintained by: Jaroslav Kysela <perex@perex.cz> ** ** This driver has only been tested with ** -- HP J2585B 10/100 Mbit/s PCI Busmaster @@ -2951,7 +2951,7 @@ static struct pci_driver hp100_pci_driver = { */ MODULE_LICENSE("GPL"); -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>, " +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, " "Siegfried \"Frieder\" Loeffler (dg1sek) <floeff@mathematik.uni-stuttgart.de>"); MODULE_DESCRIPTION("HP CASCADE Architecture Driver for 100VG-AnyLan Network Adapters"); diff --git a/drivers/pnp/interface.c b/drivers/pnp/interface.c index a0cfb75bbb8d..e0ee28a88da3 100644 --- a/drivers/pnp/interface.c +++ b/drivers/pnp/interface.c @@ -1,7 +1,7 @@ /* * interface.c - contains everything related to the user interface * - * Some code, especially possible resource dumping is based on isapnp_proc.c (c) Jaroslav Kysela <perex@suse.cz> + * Some code, especially possible resource dumping is based on isapnp_proc.c (c) Jaroslav Kysela <perex@perex.cz> * Copyright 2002 Adam Belay <ambx1@neo.rr.com> */ diff --git a/drivers/pnp/isapnp/core.c b/drivers/pnp/isapnp/core.c index b035d60a1dcc..2c925b7cd93e 100644 --- a/drivers/pnp/isapnp/core.c +++ b/drivers/pnp/isapnp/core.c @@ -1,6 +1,6 @@ /* * ISA Plug & Play support - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -53,7 +53,7 @@ static int isapnp_rdp; /* Read Data Port */ static int isapnp_reset = 1; /* reset all PnP cards (deactivate) */ static int isapnp_verbose = 1; /* verbose mode */ -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Generic ISA Plug & Play support"); module_param(isapnp_disable, int, 0); MODULE_PARM_DESC(isapnp_disable, "ISA Plug & Play disable"); diff --git a/drivers/pnp/isapnp/proc.c b/drivers/pnp/isapnp/proc.c index 560ccb640816..2b8266c3d40f 100644 --- a/drivers/pnp/isapnp/proc.c +++ b/drivers/pnp/isapnp/proc.c @@ -1,6 +1,6 @@ /* * ISA Plug & Play support - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * 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 diff --git a/drivers/pnp/manager.c b/drivers/pnp/manager.c index 0826287eef53..ea3eac2404ca 100644 --- a/drivers/pnp/manager.c +++ b/drivers/pnp/manager.c @@ -1,7 +1,7 @@ /* * manager.c - Resource Management, Conflict Resolution, Activation and Disabling of Devices * - * based on isapnp.c resource management (c) Jaroslav Kysela <perex@suse.cz> + * based on isapnp.c resource management (c) Jaroslav Kysela <perex@perex.cz> * Copyright 2003 Adam Belay <ambx1@neo.rr.com> */ diff --git a/drivers/pnp/resource.c b/drivers/pnp/resource.c index ef1286900db3..087fed18628f 100644 --- a/drivers/pnp/resource.c +++ b/drivers/pnp/resource.c @@ -1,7 +1,7 @@ /* * resource.c - Contains functions for registering and analyzing resource information * - * based on isapnp.c resource management (c) Jaroslav Kysela <perex@suse.cz> + * based on isapnp.c resource management (c) Jaroslav Kysela <perex@perex.cz> * Copyright 2003 Adam Belay <ambx1@neo.rr.com> */ diff --git a/include/linux/i2c-id.h b/include/linux/i2c-id.h index a271b67a8e2d..88c81403eb3f 100644 --- a/include/linux/i2c-id.h +++ b/include/linux/i2c-id.h @@ -120,6 +120,7 @@ #define I2C_DRIVERID_WM8753 91 /* Wolfson WM8753 audio codec */ #define I2C_DRIVERID_LM4857 92 /* LM4857 Audio Amplifier */ #define I2C_DRIVERID_VP27SMPX 93 /* Panasonic VP27s tuner internal MPX */ +#define I2C_DRIVERID_CS4270 94 /* Cirrus Logic 4270 audio codec */ #define I2C_DRIVERID_I2CDEV 900 #define I2C_DRIVERID_ARP 902 /* SMBus ARP Client */ diff --git a/include/linux/spi/at73c213.h b/include/linux/spi/at73c213.h new file mode 100644 index 000000000000..0f20a70e5eb4 --- /dev/null +++ b/include/linux/spi/at73c213.h @@ -0,0 +1,25 @@ +/* + * Board-specific data used to set up AT73c213 audio DAC driver. + */ + +#ifndef __LINUX_SPI_AT73C213_H +#define __LINUX_SPI_AT73C213_H + +/** + * at73c213_board_info - how the external DAC is wired to the device. + * + * @ssc_id: SSC platform_driver id the DAC shall use to stream the audio. + * @dac_clk: the external clock used to provide master clock to the DAC. + * @shortname: a short discription for the DAC, seen by userspace tools. + * + * This struct contains the configuration of the hardware connection to the + * external DAC. The DAC needs a master clock and a I2S audio stream. It also + * provides a name which is used to identify it in userspace tools. + */ +struct at73c213_board_info { + int ssc_id; + struct clk *dac_clk; + char shortname[32]; +}; + +#endif /* __LINUX_SPI_AT73C213_H */ diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index 246ac23534bd..01480581f825 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -2,7 +2,7 @@ #define __SOUND_AC97_CODEC_H /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Universal interface for Audio Codec '97 * * For more details look to AC '97 component specification revision 2.1 @@ -345,9 +345,9 @@ #define AC97_ALC650_GPIO_STATUS 0x78 #define AC97_ALC650_CLOCK 0x7a -/* specific - Yamaha YMF753 */ -#define AC97_YMF753_DIT_CTRL2 0x66 /* DIT Control 2 */ -#define AC97_YMF753_3D_MODE_SEL 0x68 /* 3D Mode Select */ +/* specific - Yamaha YMF7x3 */ +#define AC97_YMF7X3_DIT_CTRL 0x66 /* DIT Control (YMF743) / 2 (YMF753) */ +#define AC97_YMF7X3_3D_MODE_SEL 0x68 /* 3D Mode Select */ /* specific - C-Media */ #define AC97_CM9738_VENDOR_CTRL 0x5a diff --git a/include/sound/ad1848.h b/include/sound/ad1848.h index b2c3f00a9b35..d04f9e78c7c1 100644 --- a/include/sound/ad1848.h +++ b/include/sound/ad1848.h @@ -2,7 +2,7 @@ #define __SOUND_AD1848_H /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Definitions for AD1847/AD1848/CS4248 chips * * @@ -27,7 +27,7 @@ /* IO ports */ -#define AD1848P( codec, x ) ( (chip) -> port + c_d_c_AD1848##x ) +#define AD1848P( chip, x ) ( (chip) -> port + c_d_c_AD1848##x ) #define c_d_c_AD1848REGSEL 0 #define c_d_c_AD1848REG 1 @@ -154,7 +154,6 @@ struct snd_ad1848 { #endif spinlock_t reg_lock; - struct mutex open_mutex; }; /* exported functions */ diff --git a/include/sound/ainstr_gf1.h b/include/sound/ainstr_gf1.h index 47726fe0f46d..b62b665c69c6 100644 --- a/include/sound/ainstr_gf1.h +++ b/include/sound/ainstr_gf1.h @@ -2,7 +2,7 @@ * Advanced Linux Sound Architecture * * GF1 (GUS) Patch Instrument Format - * Copyright (c) 1994-99 by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) 1994-99 by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/include/sound/ainstr_iw.h b/include/sound/ainstr_iw.h index 251feaf1b388..11bd25082600 100644 --- a/include/sound/ainstr_iw.h +++ b/include/sound/ainstr_iw.h @@ -2,7 +2,7 @@ * Advanced Linux Sound Architecture * * InterWave FFFF Instrument Format - * Copyright (c) 1994-99 by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) 1994-99 by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/include/sound/ainstr_simple.h b/include/sound/ainstr_simple.h index 5eead12e58ae..da08e7287557 100644 --- a/include/sound/ainstr_simple.h +++ b/include/sound/ainstr_simple.h @@ -2,7 +2,7 @@ * Advanced Linux Sound Architecture * * Simple (MOD player) Instrument Format - * Copyright (c) 1994-99 by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) 1994-99 by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/include/sound/ak4114.h b/include/sound/ak4114.h index d647dae912b9..4e80d3fe7381 100644 --- a/include/sound/ak4114.h +++ b/include/sound/ak4114.h @@ -3,7 +3,7 @@ /* * Routines for Asahi Kasei AK4114 - * Copyright (c) by Jaroslav Kysela <perex@suse.cz>, + * Copyright (c) by Jaroslav Kysela <perex@perex.cz>, * * * This program is free software; you can redistribute it and/or modify diff --git a/include/sound/ak4117.h b/include/sound/ak4117.h index d650d52e3d29..1e8178171baf 100644 --- a/include/sound/ak4117.h +++ b/include/sound/ak4117.h @@ -3,7 +3,7 @@ /* * Routines for Asahi Kasei AK4117 - * Copyright (c) by Jaroslav Kysela <perex@suse.cz>, + * Copyright (c) by Jaroslav Kysela <perex@perex.cz>, * * * This program is free software; you can redistribute it and/or modify diff --git a/include/sound/ak4531_codec.h b/include/sound/ak4531_codec.h index fb30faab43a8..575296cf7987 100644 --- a/include/sound/ak4531_codec.h +++ b/include/sound/ak4531_codec.h @@ -2,7 +2,7 @@ #define __SOUND_AK4531_CODEC_H /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Universal interface for Audio Codec '97 * * For more details look to AC '97 component specification revision 2.1 diff --git a/include/sound/ak4xxx-adda.h b/include/sound/ak4xxx-adda.h index fd0a6c46f497..891cf1aea8b1 100644 --- a/include/sound/ak4xxx-adda.h +++ b/include/sound/ak4xxx-adda.h @@ -5,7 +5,7 @@ * ALSA driver for AK4524 / AK4528 / AK4529 / AK4355 / AK4381 * AD and DA converters * - * Copyright (c) 2000 Jaroslav Kysela <perex@suse.cz> + * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz> * * 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 diff --git a/include/sound/asequencer.h b/include/sound/asequencer.h index 3f2f4042a20d..64daccbe8b29 100644 --- a/include/sound/asequencer.h +++ b/include/sound/asequencer.h @@ -1,7 +1,7 @@ /* * Main header file for the ALSA sequencer * Copyright (c) 1998-1999 by Frank van de Pol <fvdpol@coil.demon.nl> - * (c) 1998-1999 by Jaroslav Kysela <perex@suse.cz> + * (c) 1998-1999 by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/include/sound/asound.h b/include/sound/asound.h index c1621c650a9a..af9d11d315e9 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -1,6 +1,6 @@ /* * Advanced Linux Sound Architecture - ALSA - Driver - * Copyright (c) 1994-2003 by Jaroslav Kysela <perex@suse.cz>, + * Copyright (c) 1994-2003 by Jaroslav Kysela <perex@perex.cz>, * Abramo Bagnara <abramo@alsa-project.org> * * @@ -92,6 +92,7 @@ enum { SNDRV_HWDEP_IFACE_USX2Y_PCM, /* Tascam US122, US224 & US428 rawusb pcm */ SNDRV_HWDEP_IFACE_PCXHR, /* Digigram PCXHR */ SNDRV_HWDEP_IFACE_SB_RC, /* SB Extigy/Audigy2NX remote control */ + SNDRV_HWDEP_IFACE_HDA, /* HD-audio */ /* Don't forget to change the following: */ SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_SB_RC diff --git a/include/sound/asound_fm.h b/include/sound/asound_fm.h index 956fdc23c595..8fbcab7cc73b 100644 --- a/include/sound/asound_fm.h +++ b/include/sound/asound_fm.h @@ -5,7 +5,7 @@ * Advanced Linux Sound Architecture - ALSA * * Interface file between ALSA driver & user space - * Copyright (c) 1994-98 by Jaroslav Kysela <perex@suse.cz>, + * Copyright (c) 1994-98 by Jaroslav Kysela <perex@perex.cz>, * 4Front Technologies * * Direct FM control diff --git a/include/sound/asoundef.h b/include/sound/asoundef.h index 58c9ef3d1825..024ce62f7d16 100644 --- a/include/sound/asoundef.h +++ b/include/sound/asoundef.h @@ -3,7 +3,7 @@ /* * Advanced Linux Sound Architecture - ALSA - Driver - * Copyright (c) 1994-2000 by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) 1994-2000 by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/include/sound/control.h b/include/sound/control.h index 72e759f619b1..e79baa63912f 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -3,7 +3,7 @@ /* * Header file for control interface - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -161,4 +161,12 @@ static inline struct snd_ctl_elem_id *snd_ctl_build_ioff(struct snd_ctl_elem_id return dst_id; } +/* + * Frequently used control callbacks + */ +int snd_ctl_boolean_mono_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +int snd_ctl_boolean_stereo_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); + #endif /* __SOUND_CONTROL_H */ diff --git a/include/sound/core.h b/include/sound/core.h index 4b9e609975ab..6954836487ed 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -3,7 +3,7 @@ /* * Main header file for the ALSA driver - * Copyright (c) 1994-2001 by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) 1994-2001 by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/include/sound/cs4231-regs.h b/include/sound/cs4231-regs.h new file mode 100644 index 000000000000..f1490265c9b8 --- /dev/null +++ b/include/sound/cs4231-regs.h @@ -0,0 +1,180 @@ +#ifndef __SOUND_CS4231_REGS_H +#define __SOUND_CS4231_REGS_H + +/* + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> + * Definitions for CS4231 & InterWave chips & compatible chips registers + * + * + * 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 + * + */ + +/* IO ports */ + +#define CS4231P(x) (c_d_c_CS4231##x) + +#define c_d_c_CS4231REGSEL 0 +#define c_d_c_CS4231REG 1 +#define c_d_c_CS4231STATUS 2 +#define c_d_c_CS4231PIO 3 + +/* codec registers */ + +#define CS4231_LEFT_INPUT 0x00 /* left input control */ +#define CS4231_RIGHT_INPUT 0x01 /* right input control */ +#define CS4231_AUX1_LEFT_INPUT 0x02 /* left AUX1 input control */ +#define CS4231_AUX1_RIGHT_INPUT 0x03 /* right AUX1 input control */ +#define CS4231_AUX2_LEFT_INPUT 0x04 /* left AUX2 input control */ +#define CS4231_AUX2_RIGHT_INPUT 0x05 /* right AUX2 input control */ +#define CS4231_LEFT_OUTPUT 0x06 /* left output control register */ +#define CS4231_RIGHT_OUTPUT 0x07 /* right output control register */ +#define CS4231_PLAYBK_FORMAT 0x08 /* clock and data format - playback - bits 7-0 MCE */ +#define CS4231_IFACE_CTRL 0x09 /* interface control - bits 7-2 MCE */ +#define CS4231_PIN_CTRL 0x0a /* pin control */ +#define CS4231_TEST_INIT 0x0b /* test and initialization */ +#define CS4231_MISC_INFO 0x0c /* miscellaneaous information */ +#define CS4231_LOOPBACK 0x0d /* loopback control */ +#define CS4231_PLY_UPR_CNT 0x0e /* playback upper base count */ +#define CS4231_PLY_LWR_CNT 0x0f /* playback lower base count */ +#define CS4231_ALT_FEATURE_1 0x10 /* alternate #1 feature enable */ +#define AD1845_AF1_MIC_LEFT 0x10 /* alternate #1 feature + MIC left */ +#define CS4231_ALT_FEATURE_2 0x11 /* alternate #2 feature enable */ +#define AD1845_AF2_MIC_RIGHT 0x11 /* alternate #2 feature + MIC right */ +#define CS4231_LEFT_LINE_IN 0x12 /* left line input control */ +#define CS4231_RIGHT_LINE_IN 0x13 /* right line input control */ +#define CS4231_TIMER_LOW 0x14 /* timer low byte */ +#define CS4231_TIMER_HIGH 0x15 /* timer high byte */ +#define CS4231_LEFT_MIC_INPUT 0x16 /* left MIC input control register (InterWave only) */ +#define AD1845_UPR_FREQ_SEL 0x16 /* upper byte of frequency select */ +#define CS4231_RIGHT_MIC_INPUT 0x17 /* right MIC input control register (InterWave only) */ +#define AD1845_LWR_FREQ_SEL 0x17 /* lower byte of frequency select */ +#define CS4236_EXT_REG 0x17 /* extended register access */ +#define CS4231_IRQ_STATUS 0x18 /* irq status register */ +#define CS4231_LINE_LEFT_OUTPUT 0x19 /* left line output control register (InterWave only) */ +#define CS4231_VERSION 0x19 /* CS4231(A) - version values */ +#define CS4231_MONO_CTRL 0x1a /* mono input/output control */ +#define CS4231_LINE_RIGHT_OUTPUT 0x1b /* right line output control register (InterWave only) */ +#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 */ +#define CS4231_REC_LWR_CNT 0x1f /* record lower count */ + +/* definitions for codec register select port - CODECP( REGSEL ) */ + +#define CS4231_INIT 0x80 /* CODEC is initializing */ +#define CS4231_MCE 0x40 /* mode change enable */ +#define CS4231_TRD 0x20 /* transfer request disable */ + +/* definitions for codec status register - CODECP( STATUS ) */ + +#define CS4231_GLOBALIRQ 0x01 /* IRQ is active */ + +/* definitions for codec irq status */ + +#define CS4231_PLAYBACK_IRQ 0x10 +#define CS4231_RECORD_IRQ 0x20 +#define CS4231_TIMER_IRQ 0x40 +#define CS4231_ALL_IRQS 0x70 +#define CS4231_REC_UNDERRUN 0x08 +#define CS4231_REC_OVERRUN 0x04 +#define CS4231_PLY_OVERRUN 0x02 +#define CS4231_PLY_UNDERRUN 0x01 + +/* definitions for CS4231_LEFT_INPUT and CS4231_RIGHT_INPUT registers */ + +#define CS4231_ENABLE_MIC_GAIN 0x20 + +#define CS4231_MIXS_LINE 0x00 +#define CS4231_MIXS_AUX1 0x40 +#define CS4231_MIXS_MIC 0x80 +#define CS4231_MIXS_ALL 0xc0 + +/* definitions for clock and data format register - CS4231_PLAYBK_FORMAT */ + +#define CS4231_LINEAR_8 0x00 /* 8-bit unsigned data */ +#define CS4231_ALAW_8 0x60 /* 8-bit A-law companded */ +#define CS4231_ULAW_8 0x20 /* 8-bit U-law companded */ +#define CS4231_LINEAR_16 0x40 /* 16-bit twos complement data - little endian */ +#define CS4231_LINEAR_16_BIG 0xc0 /* 16-bit twos complement data - big endian */ +#define CS4231_ADPCM_16 0xa0 /* 16-bit ADPCM */ +#define CS4231_STEREO 0x10 /* stereo mode */ +/* bits 3-1 define frequency divisor */ +#define CS4231_XTAL1 0x00 /* 24.576 crystal */ +#define CS4231_XTAL2 0x01 /* 16.9344 crystal */ + +/* definitions for interface control register - CS4231_IFACE_CTRL */ + +#define CS4231_RECORD_PIO 0x80 /* record PIO enable */ +#define CS4231_PLAYBACK_PIO 0x40 /* playback PIO enable */ +#define CS4231_CALIB_MODE 0x18 /* calibration mode bits */ +#define CS4231_AUTOCALIB 0x08 /* auto calibrate */ +#define CS4231_SINGLE_DMA 0x04 /* use single DMA channel */ +#define CS4231_RECORD_ENABLE 0x02 /* record enable */ +#define CS4231_PLAYBACK_ENABLE 0x01 /* playback enable */ + +/* definitions for pin control register - CS4231_PIN_CTRL */ + +#define CS4231_IRQ_ENABLE 0x02 /* enable IRQ */ +#define CS4231_XCTL1 0x40 /* external control #1 */ +#define CS4231_XCTL0 0x80 /* external control #0 */ + +/* definitions for test and init register - CS4231_TEST_INIT */ + +#define CS4231_CALIB_IN_PROGRESS 0x20 /* auto calibrate in progress */ +#define CS4231_DMA_REQUEST 0x10 /* DMA request in progress */ + +/* definitions for misc control register - CS4231_MISC_INFO */ + +#define CS4231_MODE2 0x40 /* MODE 2 */ +#define CS4231_IW_MODE3 0x6c /* MODE 3 - InterWave enhanced mode */ +#define CS4231_4236_MODE3 0xe0 /* MODE 3 - CS4236+ enhanced mode */ + +/* definitions for alternate feature 1 register - CS4231_ALT_FEATURE_1 */ + +#define CS4231_DACZ 0x01 /* zero DAC when underrun */ +#define CS4231_TIMER_ENABLE 0x40 /* codec timer enable */ +#define CS4231_OLB 0x80 /* output level bit */ + +/* definitions for Extended Registers - CS4236+ */ + +#define CS4236_REG(i23val) (((i23val << 2) & 0x10) | ((i23val >> 4) & 0x0f)) +#define CS4236_I23VAL(reg) ((((reg)&0xf) << 4) | (((reg)&0x10) >> 2) | 0x8) + +#define CS4236_LEFT_LINE 0x08 /* left LINE alternate volume */ +#define CS4236_RIGHT_LINE 0x18 /* right LINE alternate volume */ +#define CS4236_LEFT_MIC 0x28 /* left MIC volume */ +#define CS4236_RIGHT_MIC 0x38 /* right MIC volume */ +#define CS4236_LEFT_MIX_CTRL 0x48 /* synthesis and left input mixer control */ +#define CS4236_RIGHT_MIX_CTRL 0x58 /* right input mixer control */ +#define CS4236_LEFT_FM 0x68 /* left FM volume */ +#define CS4236_RIGHT_FM 0x78 /* right FM volume */ +#define CS4236_LEFT_DSP 0x88 /* left DSP serial port volume */ +#define CS4236_RIGHT_DSP 0x98 /* right DSP serial port volume */ +#define CS4236_RIGHT_LOOPBACK 0xa8 /* right loopback monitor volume */ +#define CS4236_DAC_MUTE 0xb8 /* DAC mute and IFSE enable */ +#define CS4236_ADC_RATE 0xc8 /* indenpendent ADC sample frequency */ +#define CS4236_DAC_RATE 0xd8 /* indenpendent DAC sample frequency */ +#define CS4236_LEFT_MASTER 0xe8 /* left master digital audio volume */ +#define CS4236_RIGHT_MASTER 0xf8 /* right master digital audio volume */ +#define CS4236_LEFT_WAVE 0x0c /* left wavetable serial port volume */ +#define CS4236_RIGHT_WAVE 0x1c /* right wavetable serial port volume */ +#define CS4236_VERSION 0x9c /* chip version and ID */ + +#endif /* __SOUND_CS4231_REGS_H */ diff --git a/include/sound/cs4231.h b/include/sound/cs4231.h index ab51ce1ba9a5..66055d702aa3 100644 --- a/include/sound/cs4231.h +++ b/include/sound/cs4231.h @@ -2,7 +2,7 @@ #define __SOUND_CS4231_H /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Definitions for CS4231 & InterWave chips & compatible chips * * @@ -26,160 +26,7 @@ #include "pcm.h" #include "timer.h" -/* IO ports */ - -#define CS4231P(x) (c_d_c_CS4231##x) - -#define c_d_c_CS4231REGSEL 0 -#define c_d_c_CS4231REG 1 -#define c_d_c_CS4231STATUS 2 -#define c_d_c_CS4231PIO 3 - -/* codec registers */ - -#define CS4231_LEFT_INPUT 0x00 /* left input control */ -#define CS4231_RIGHT_INPUT 0x01 /* right input control */ -#define CS4231_AUX1_LEFT_INPUT 0x02 /* left AUX1 input control */ -#define CS4231_AUX1_RIGHT_INPUT 0x03 /* right AUX1 input control */ -#define CS4231_AUX2_LEFT_INPUT 0x04 /* left AUX2 input control */ -#define CS4231_AUX2_RIGHT_INPUT 0x05 /* right AUX2 input control */ -#define CS4231_LEFT_OUTPUT 0x06 /* left output control register */ -#define CS4231_RIGHT_OUTPUT 0x07 /* right output control register */ -#define CS4231_PLAYBK_FORMAT 0x08 /* clock and data format - playback - bits 7-0 MCE */ -#define CS4231_IFACE_CTRL 0x09 /* interface control - bits 7-2 MCE */ -#define CS4231_PIN_CTRL 0x0a /* pin control */ -#define CS4231_TEST_INIT 0x0b /* test and initialization */ -#define CS4231_MISC_INFO 0x0c /* miscellaneaous information */ -#define CS4231_LOOPBACK 0x0d /* loopback control */ -#define CS4231_PLY_UPR_CNT 0x0e /* playback upper base count */ -#define CS4231_PLY_LWR_CNT 0x0f /* playback lower base count */ -#define CS4231_ALT_FEATURE_1 0x10 /* alternate #1 feature enable */ -#define AD1845_AF1_MIC_LEFT 0x10 /* alternate #1 feature + MIC left */ -#define CS4231_ALT_FEATURE_2 0x11 /* alternate #2 feature enable */ -#define AD1845_AF2_MIC_RIGHT 0x11 /* alternate #2 feature + MIC right */ -#define CS4231_LEFT_LINE_IN 0x12 /* left line input control */ -#define CS4231_RIGHT_LINE_IN 0x13 /* right line input control */ -#define CS4231_TIMER_LOW 0x14 /* timer low byte */ -#define CS4231_TIMER_HIGH 0x15 /* timer high byte */ -#define CS4231_LEFT_MIC_INPUT 0x16 /* left MIC input control register (InterWave only) */ -#define AD1845_UPR_FREQ_SEL 0x16 /* upper byte of frequency select */ -#define CS4231_RIGHT_MIC_INPUT 0x17 /* right MIC input control register (InterWave only) */ -#define AD1845_LWR_FREQ_SEL 0x17 /* lower byte of frequency select */ -#define CS4236_EXT_REG 0x17 /* extended register access */ -#define CS4231_IRQ_STATUS 0x18 /* irq status register */ -#define CS4231_LINE_LEFT_OUTPUT 0x19 /* left line output control register (InterWave only) */ -#define CS4231_VERSION 0x19 /* CS4231(A) - version values */ -#define CS4231_MONO_CTRL 0x1a /* mono input/output control */ -#define CS4231_LINE_RIGHT_OUTPUT 0x1b /* right line output control register (InterWave only) */ -#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 */ -#define CS4231_REC_LWR_CNT 0x1f /* record lower count */ - -/* definitions for codec register select port - CODECP( REGSEL ) */ - -#define CS4231_INIT 0x80 /* CODEC is initializing */ -#define CS4231_MCE 0x40 /* mode change enable */ -#define CS4231_TRD 0x20 /* transfer request disable */ - -/* definitions for codec status register - CODECP( STATUS ) */ - -#define CS4231_GLOBALIRQ 0x01 /* IRQ is active */ - -/* definitions for codec irq status */ - -#define CS4231_PLAYBACK_IRQ 0x10 -#define CS4231_RECORD_IRQ 0x20 -#define CS4231_TIMER_IRQ 0x40 -#define CS4231_ALL_IRQS 0x70 -#define CS4231_REC_UNDERRUN 0x08 -#define CS4231_REC_OVERRUN 0x04 -#define CS4231_PLY_OVERRUN 0x02 -#define CS4231_PLY_UNDERRUN 0x01 - -/* definitions for CS4231_LEFT_INPUT and CS4231_RIGHT_INPUT registers */ - -#define CS4231_ENABLE_MIC_GAIN 0x20 - -#define CS4231_MIXS_LINE 0x00 -#define CS4231_MIXS_AUX1 0x40 -#define CS4231_MIXS_MIC 0x80 -#define CS4231_MIXS_ALL 0xc0 - -/* definitions for clock and data format register - CS4231_PLAYBK_FORMAT */ - -#define CS4231_LINEAR_8 0x00 /* 8-bit unsigned data */ -#define CS4231_ALAW_8 0x60 /* 8-bit A-law companded */ -#define CS4231_ULAW_8 0x20 /* 8-bit U-law companded */ -#define CS4231_LINEAR_16 0x40 /* 16-bit twos complement data - little endian */ -#define CS4231_LINEAR_16_BIG 0xc0 /* 16-bit twos complement data - big endian */ -#define CS4231_ADPCM_16 0xa0 /* 16-bit ADPCM */ -#define CS4231_STEREO 0x10 /* stereo mode */ -/* bits 3-1 define frequency divisor */ -#define CS4231_XTAL1 0x00 /* 24.576 crystal */ -#define CS4231_XTAL2 0x01 /* 16.9344 crystal */ - -/* definitions for interface control register - CS4231_IFACE_CTRL */ - -#define CS4231_RECORD_PIO 0x80 /* record PIO enable */ -#define CS4231_PLAYBACK_PIO 0x40 /* playback PIO enable */ -#define CS4231_CALIB_MODE 0x18 /* calibration mode bits */ -#define CS4231_AUTOCALIB 0x08 /* auto calibrate */ -#define CS4231_SINGLE_DMA 0x04 /* use single DMA channel */ -#define CS4231_RECORD_ENABLE 0x02 /* record enable */ -#define CS4231_PLAYBACK_ENABLE 0x01 /* playback enable */ - -/* definitions for pin control register - CS4231_PIN_CTRL */ - -#define CS4231_IRQ_ENABLE 0x02 /* enable IRQ */ -#define CS4231_XCTL1 0x40 /* external control #1 */ -#define CS4231_XCTL0 0x80 /* external control #0 */ - -/* definitions for test and init register - CS4231_TEST_INIT */ - -#define CS4231_CALIB_IN_PROGRESS 0x20 /* auto calibrate in progress */ -#define CS4231_DMA_REQUEST 0x10 /* DMA request in progress */ - -/* definitions for misc control register - CS4231_MISC_INFO */ - -#define CS4231_MODE2 0x40 /* MODE 2 */ -#define CS4231_IW_MODE3 0x6c /* MODE 3 - InterWave enhanced mode */ -#define CS4231_4236_MODE3 0xe0 /* MODE 3 - CS4236+ enhanced mode */ - -/* definitions for alternate feature 1 register - CS4231_ALT_FEATURE_1 */ - -#define CS4231_DACZ 0x01 /* zero DAC when underrun */ -#define CS4231_TIMER_ENABLE 0x40 /* codec timer enable */ -#define CS4231_OLB 0x80 /* output level bit */ - -/* definitions for Extended Registers - CS4236+ */ - -#define CS4236_REG(i23val) (((i23val << 2) & 0x10) | ((i23val >> 4) & 0x0f)) -#define CS4236_I23VAL(reg) ((((reg)&0xf) << 4) | (((reg)&0x10) >> 2) | 0x8) - -#define CS4236_LEFT_LINE 0x08 /* left LINE alternate volume */ -#define CS4236_RIGHT_LINE 0x18 /* right LINE alternate volume */ -#define CS4236_LEFT_MIC 0x28 /* left MIC volume */ -#define CS4236_RIGHT_MIC 0x38 /* right MIC volume */ -#define CS4236_LEFT_MIX_CTRL 0x48 /* synthesis and left input mixer control */ -#define CS4236_RIGHT_MIX_CTRL 0x58 /* right input mixer control */ -#define CS4236_LEFT_FM 0x68 /* left FM volume */ -#define CS4236_RIGHT_FM 0x78 /* right FM volume */ -#define CS4236_LEFT_DSP 0x88 /* left DSP serial port volume */ -#define CS4236_RIGHT_DSP 0x98 /* right DSP serial port volume */ -#define CS4236_RIGHT_LOOPBACK 0xa8 /* right loopback monitor volume */ -#define CS4236_DAC_MUTE 0xb8 /* DAC mute and IFSE enable */ -#define CS4236_ADC_RATE 0xc8 /* indenpendent ADC sample frequency */ -#define CS4236_DAC_RATE 0xd8 /* indenpendent DAC sample frequency */ -#define CS4236_LEFT_MASTER 0xe8 /* left master digital audio volume */ -#define CS4236_RIGHT_MASTER 0xf8 /* right master digital audio volume */ -#define CS4236_LEFT_WAVE 0x0c /* left wavetable serial port volume */ -#define CS4236_RIGHT_WAVE 0x1c /* right wavetable serial port volume */ -#define CS4236_VERSION 0x9c /* chip version and ID */ +#include "cs4231-regs.h" /* defines for codec.mode */ @@ -210,7 +57,7 @@ #define CS4231_HW_CS4239 0x0404 /* CS4239 - Crystal Clear (tm) stereo enhancement */ /* compatible, but clones */ #define CS4231_HW_INTERWAVE 0x1000 /* InterWave chip */ -#define CS4231_HW_OPL3SA2 0x1001 /* OPL3-SA2 chip */ +#define CS4231_HW_OPL3SA2 0x1101 /* OPL3-SA2 chip, similar to cs4231 */ /* defines for codec.hwshare */ #define CS4231_HWSHARE_IRQ (1<<0) diff --git a/include/sound/cs46xx.h b/include/sound/cs46xx.h index 353910ce9755..6b40ee60f4c5 100644 --- a/include/sound/cs46xx.h +++ b/include/sound/cs46xx.h @@ -2,7 +2,7 @@ #define __SOUND_CS46XX_H /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz>, + * Copyright (c) by Jaroslav Kysela <perex@perex.cz>, * Cirrus Logic, Inc. * Definitions for Cirrus Logic CS46xx chips * diff --git a/include/sound/cs46xx_dsp_scb_types.h b/include/sound/cs46xx_dsp_scb_types.h index 9cb6c7d09567..080857ad0ca2 100644 --- a/include/sound/cs46xx_dsp_scb_types.h +++ b/include/sound/cs46xx_dsp_scb_types.h @@ -1,6 +1,6 @@ /* * The driver for the Cirrus Logic's Sound Fusion CS46XX based soundcards - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/include/sound/cs46xx_dsp_spos.h b/include/sound/cs46xx_dsp_spos.h index d9da9e59cf37..7c44667e79a6 100644 --- a/include/sound/cs46xx_dsp_spos.h +++ b/include/sound/cs46xx_dsp_spos.h @@ -1,6 +1,6 @@ /* * The driver for the Cirrus Logic's Sound Fusion CS46XX based soundcards - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/include/sound/cs46xx_dsp_task_types.h b/include/sound/cs46xx_dsp_task_types.h index b3076c487de6..5cf920bfda27 100644 --- a/include/sound/cs46xx_dsp_task_types.h +++ b/include/sound/cs46xx_dsp_task_types.h @@ -1,6 +1,6 @@ /* * The driver for the Cirrus Logic's Sound Fusion CS46XX based soundcards - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/include/sound/cs8403.h b/include/sound/cs8403.h index c6c3f9f0da78..3a8c174a4209 100644 --- a/include/sound/cs8403.h +++ b/include/sound/cs8403.h @@ -3,7 +3,7 @@ /* * Routines for Cirrus Logic CS8403/CS8404A IEC958 (S/PDIF) Transmitter - * Copyright (c) by Jaroslav Kysela <perex@suse.cz>, + * Copyright (c) by Jaroslav Kysela <perex@perex.cz>, * Takashi Iwai <tiwai@suse.de> * * diff --git a/include/sound/cs8427.h b/include/sound/cs8427.h index 97fd9acf8028..f862cfff5f6a 100644 --- a/include/sound/cs8427.h +++ b/include/sound/cs8427.h @@ -3,7 +3,7 @@ /* * Routines for Cirrus Logic CS8427 - * Copyright (c) by Jaroslav Kysela <perex@suse.cz>, + * Copyright (c) by Jaroslav Kysela <perex@perex.cz>, * * * This program is free software; you can redistribute it and/or modify diff --git a/include/sound/driver.h b/include/sound/driver.h index 3c522e59a33c..5ccb6c5feecb 100644 --- a/include/sound/driver.h +++ b/include/sound/driver.h @@ -3,7 +3,7 @@ /* * Main header file for the ALSA driver - * Copyright (c) 1994-2000 by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) 1994-2000 by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 529d0a564367..441aa06dcd6f 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -2,7 +2,7 @@ #define __SOUND_EMU10K1_H /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz>, + * Copyright (c) by Jaroslav Kysela <perex@perex.cz>, * Creative Labs, Inc. * Definitions for EMU10K1 (SB Live!) chips * @@ -1408,8 +1408,6 @@ struct snd_emu10k1_fx8010 { struct snd_emu10k1_fx8010_irq *irq_handlers; }; -#define emu10k1_gpr_ctl(n) list_entry(n, struct snd_emu10k1_fx8010_ctl, list) - struct snd_emu10k1_midi { struct snd_emu10k1 *emu; struct snd_rawmidi *rmidi; @@ -1456,6 +1454,9 @@ struct snd_emu1010 { unsigned int adc_pads; /* bit mask */ unsigned int dac_pads; /* bit mask */ unsigned int internal_clock; /* 44100 or 48000 */ + unsigned int optical_in; /* 0:SPDIF, 1:ADAT */ + unsigned int optical_out; /* 0:SPDIF, 1:ADAT */ + struct task_struct *firmware_thread; }; struct snd_emu10k1 { @@ -1599,9 +1600,9 @@ unsigned int snd_emu10k1_ptr20_read(struct snd_emu10k1 * emu, unsigned int reg, void snd_emu10k1_ptr20_write(struct snd_emu10k1 *emu, unsigned int reg, unsigned int chn, unsigned int data); int snd_emu10k1_spi_write(struct snd_emu10k1 * emu, unsigned int data); int snd_emu10k1_i2c_write(struct snd_emu10k1 *emu, u32 reg, u32 value); -int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, int reg, int value); -int snd_emu1010_fpga_read(struct snd_emu10k1 * emu, int reg, int *value); -int snd_emu1010_fpga_link_dst_src_write(struct snd_emu10k1 * emu, int dst, int src); +int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, u32 reg, u32 value); +int snd_emu1010_fpga_read(struct snd_emu10k1 * emu, u32 reg, u32 *value); +int snd_emu1010_fpga_link_dst_src_write(struct snd_emu10k1 * emu, u32 dst, u32 src); unsigned int snd_emu10k1_efx_read(struct snd_emu10k1 *emu, unsigned int pc); void snd_emu10k1_intr_enable(struct snd_emu10k1 *emu, unsigned int intrenb); void snd_emu10k1_intr_disable(struct snd_emu10k1 *emu, unsigned int intrenb); @@ -1746,6 +1747,8 @@ int snd_emu10k1_fx8010_unregister_irq_handler(struct snd_emu10k1 *emu, #define A_FXBUS2(x) (0x80 + (x)) /* x = 0x00 - 0x1f extra outs used for EFX capture -> A_FXWC2 */ #define A_EMU32OUTH(x) (0xa0 + (x)) /* x = 0x00 - 0x0f "EMU32_OUT_10 - _1F" - ??? */ #define A_EMU32OUTL(x) (0xb0 + (x)) /* x = 0x00 - 0x0f "EMU32_OUT_1 - _F" - ??? */ +#define A3_EMU32IN(x) (0x160 + (x)) /* x = 0x00 - 0x3f "EMU32_IN_00 - _3F" - Only when .device = 0x0008 */ +#define A3_EMU32OUT(x) (0x1E0 + (x)) /* x = 0x00 - 0x0f "EMU32_OUT_00 - _3F" - Only when .device = 0x0008 */ #define A_GPR(x) (A_FXGPREGBASE + (x)) /* cc_reg constants */ diff --git a/include/sound/es1688.h b/include/sound/es1688.h index fc1c47dae3da..10fcf1465810 100644 --- a/include/sound/es1688.h +++ b/include/sound/es1688.h @@ -3,7 +3,7 @@ /* * Header file for ES488/ES1688 - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/include/sound/gus.h b/include/sound/gus.h index c49ea57db8cc..e5433d8b78bc 100644 --- a/include/sound/gus.h +++ b/include/sound/gus.h @@ -3,7 +3,7 @@ /* * Global structures used for GUS part of ALSA driver - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/include/sound/hda_hwdep.h b/include/sound/hda_hwdep.h new file mode 100644 index 000000000000..1c0034e87f22 --- /dev/null +++ b/include/sound/hda_hwdep.h @@ -0,0 +1,44 @@ +/* + * HWDEP Interface for HD-audio codec + * + * Copyright (c) 2007 Takashi Iwai <tiwai@suse.de> + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef __SOUND_HDA_HWDEP_H +#define __SOUND_HDA_HWDEP_H + +#define HDA_HWDEP_VERSION ((1 << 16) | (0 << 8) | (0 << 0)) /* 1.0.0 */ + +/* verb */ +#define HDA_REG_NID_SHIFT 24 +#define HDA_REG_VERB_SHIFT 8 +#define HDA_REG_VAL_SHIFT 0 +#define HDA_VERB(nid,verb,param) ((nid)<<24 | (verb)<<8 | (param)) + +struct hda_verb_ioctl { + u32 verb; /* HDA_VERB() */ + u32 res; /* response */ +}; + +/* + * ioctls + */ +#define HDA_IOCTL_PVERSION _IOR('H', 0x10, int) +#define HDA_IOCTL_VERB_WRITE _IOWR('H', 0x11, struct hda_verb_ioctl) +#define HDA_IOCTL_GET_WCAP _IOWR('H', 0x12, struct hda_verb_ioctl) + +#endif diff --git a/include/sound/hdspm.h b/include/sound/hdspm.h index c3c854d99c28..81990b2bcc98 100644 --- a/include/sound/hdspm.h +++ b/include/sound/hdspm.h @@ -1,4 +1,4 @@ -#ifndef __SOUND_HDSPM_H /* -*- linux-c -*- */ +#ifndef __SOUND_HDSPM_H #define __SOUND_HDSPM_H /* * Copyright (C) 2003 Winfried Ritsch (IEM) @@ -61,7 +61,8 @@ struct hdspm_peak_rms_ioctl { }; /* use indirect access due to the limit of ioctl bit size */ -#define SNDRV_HDSPM_IOCTL_GET_PEAK_RMS _IOR('H', 0x40, struct hdspm_peak_rms_ioctl) +#define SNDRV_HDSPM_IOCTL_GET_PEAK_RMS \ + _IOR('H', 0x40, struct hdspm_peak_rms_ioctl) /* ------------ CONFIG block IOCTL ---------------------- */ @@ -79,7 +80,8 @@ struct hdspm_config_info { unsigned int analog_out; }; -#define SNDRV_HDSPM_IOCTL_GET_CONFIG_INFO _IOR('H', 0x41, struct hdspm_config_info) +#define SNDRV_HDSPM_IOCTL_GET_CONFIG_INFO \ + _IOR('H', 0x41, struct hdspm_config_info) /* get Soundcard Version */ @@ -93,10 +95,14 @@ struct hdspm_version { /* ------------- get Matrix Mixer IOCTL --------------- */ -/* MADI mixer: 64inputs+64playback in 64outputs = 8192 => *4Byte = 32768 Bytes */ +/* MADI mixer: 64inputs+64playback in 64outputs = 8192 => *4Byte = + * 32768 Bytes + */ /* organisation is 64 channelfader in a continous memory block */ -/* equivalent to hardware definition, maybe for future feature of mmap of them */ +/* equivalent to hardware definition, maybe for future feature of mmap of + * them + */ /* each of 64 outputs has 64 infader and 64 outfader: Ins to Outs mixer[out].in[in], Outstreams to Outs mixer[out].pb[pb] */ diff --git a/include/sound/hwdep.h b/include/sound/hwdep.h index 94c387b5d724..d9eea013c753 100644 --- a/include/sound/hwdep.h +++ b/include/sound/hwdep.h @@ -3,7 +3,7 @@ /* * Hardware dependent layer - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/include/sound/info.h b/include/sound/info.h index 97ffc4fb9969..fecbb1ffd540 100644 --- a/include/sound/info.h +++ b/include/sound/info.h @@ -3,7 +3,7 @@ /* * Header file for info interface - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/include/sound/initval.h b/include/sound/initval.h index e85b90750a59..1daa6dff8297 100644 --- a/include/sound/initval.h +++ b/include/sound/initval.h @@ -3,7 +3,7 @@ /* * Init values for soundcard modules - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * 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 diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index 83489c3abbaf..ae2921d9ddcc 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Takashi Iwai <tiwai@suse.de> * * Generic memory allocators diff --git a/include/sound/mixer_oss.h b/include/sound/mixer_oss.h index 197b9e3d612b..51fbcb4a277a 100644 --- a/include/sound/mixer_oss.h +++ b/include/sound/mixer_oss.h @@ -3,7 +3,7 @@ /* * OSS MIXER API - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/include/sound/mpu401.h b/include/sound/mpu401.h index d5c1396c4c9e..d45218b44dfe 100644 --- a/include/sound/mpu401.h +++ b/include/sound/mpu401.h @@ -3,7 +3,7 @@ /* * Header file for MPU-401 and compatible cards - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -50,7 +50,6 @@ #define MPU401_INFO_INTEGRATED (1 << 2) /* integrated h/w port */ #define MPU401_INFO_MMIO (1 << 3) /* MMIO access */ #define MPU401_INFO_TX_IRQ (1 << 4) /* independent TX irq */ -#define MPU401_INFO_UART_ONLY (1 << 5) /* No ENTER_UART cmd needed */ #define MPU401_MODE_BIT_INPUT 0 #define MPU401_MODE_BIT_OUTPUT 1 diff --git a/include/sound/opl3.h b/include/sound/opl3.h index 82fdb0930720..1d14b3f82393 100644 --- a/include/sound/opl3.h +++ b/include/sound/opl3.h @@ -4,7 +4,7 @@ /* * Definitions of the OPL-3 registers. * - * Copyright (c) by Jaroslav Kysela <perex@suse.cz>, + * Copyright (c) by Jaroslav Kysela <perex@perex.cz>, * Hannu Savolainen 1993-1996 * * diff --git a/include/sound/pcm-indirect.h b/include/sound/pcm-indirect.h index 7003d7702e26..1df7acaaa535 100644 --- a/include/sound/pcm-indirect.h +++ b/include/sound/pcm-indirect.h @@ -2,7 +2,7 @@ * Helper functions for indirect PCM data transfer * * Copyright (c) by Takashi Iwai <tiwai@suse.de> - * Jaroslav Kysela <perex@suse.cz> + * Jaroslav Kysela <perex@perex.cz> * * 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 diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 73334e0f823f..5e9cc460075e 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -3,7 +3,7 @@ /* * Digital Audio (PCM) abstract layer - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Abramo Bagnara <abramo@alsa-project.org> * * @@ -301,8 +301,8 @@ struct snd_pcm_runtime { union snd_pcm_sync_id sync; /* hardware synchronization ID */ /* -- mmap -- */ - volatile struct snd_pcm_mmap_status *status; - volatile struct snd_pcm_mmap_control *control; + struct snd_pcm_mmap_status *status; + struct snd_pcm_mmap_control *control; /* -- locking / scheduling -- */ wait_queue_head_t sleep; @@ -791,13 +791,13 @@ static inline struct snd_interval *hw_param_interval(struct snd_pcm_hw_params *p static inline const struct snd_mask *hw_param_mask_c(const struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var) { - return (const struct snd_mask *)hw_param_mask((struct snd_pcm_hw_params*) params, var); + return ¶ms->masks[var - SNDRV_PCM_HW_PARAM_FIRST_MASK]; } static inline const struct snd_interval *hw_param_interval_c(const struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var) { - return (const struct snd_interval *)hw_param_interval((struct snd_pcm_hw_params*) params, var); + return ¶ms->intervals[var - SNDRV_PCM_HW_PARAM_FIRST_INTERVAL]; } #define params_access(p) snd_mask_min(hw_param_mask((p), SNDRV_PCM_HW_PARAM_ACCESS)) @@ -922,7 +922,10 @@ snd_pcm_sframes_t snd_pcm_lib_writev(struct snd_pcm_substream *substream, snd_pcm_sframes_t snd_pcm_lib_readv(struct snd_pcm_substream *substream, void __user **bufs, snd_pcm_uframes_t frames); +extern const struct snd_pcm_hw_constraint_list snd_pcm_known_rates; + int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime); +unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate); static inline void snd_pcm_set_runtime_buffer(struct snd_pcm_substream *substream, struct snd_dma_buffer *bufp) diff --git a/include/sound/pcm_oss.h b/include/sound/pcm_oss.h index 1cd4f64cdf31..cc4e226f35fd 100644 --- a/include/sound/pcm_oss.h +++ b/include/sound/pcm_oss.h @@ -3,7 +3,7 @@ /* * Digital Audio (PCM) - OSS compatibility abstract layer - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index 7dbcd10fa215..b550a416d075 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -3,7 +3,7 @@ /* * Abstract layer for MIDI v1.0 stream - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/include/sound/sb.h b/include/sound/sb.h index 3ad854b397d2..d0c9ed3546c8 100644 --- a/include/sound/sb.h +++ b/include/sound/sb.h @@ -3,7 +3,7 @@ /* * Header file for SoundBlaster cards - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/include/sound/seq_instr.h b/include/sound/seq_instr.h index f2db03bfd74e..93b0c51df5b0 100644 --- a/include/sound/seq_instr.h +++ b/include/sound/seq_instr.h @@ -3,7 +3,7 @@ /* * Main kernel header file for the ALSA sequencer - * Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/include/sound/seq_midi_event.h b/include/sound/seq_midi_event.h index dd789e7cdb20..5efab8b29c57 100644 --- a/include/sound/seq_midi_event.h +++ b/include/sound/seq_midi_event.h @@ -5,7 +5,7 @@ * MIDI byte <-> sequencer event coder * * Copyright (C) 1998,99 Takashi Iwai <tiwai@suse.de>, - * Jaroslav Kysela <perex@suse.cz> + * Jaroslav Kysela <perex@perex.cz> * * 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 diff --git a/include/sound/seq_virmidi.h b/include/sound/seq_virmidi.h index 8d5aea76d7c3..d888433a3096 100644 --- a/include/sound/seq_virmidi.h +++ b/include/sound/seq_virmidi.h @@ -4,7 +4,7 @@ /* * Virtual Raw MIDI client on Sequencer * Copyright (c) 2000 by Takashi Iwai <tiwai@suse.de>, - * Jaroslav Kysela <perex@suse.cz> + * Jaroslav Kysela <perex@perex.cz> * * 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 diff --git a/include/sound/soc.h b/include/sound/soc.h index db6edba8ef08..f47ef1f75f18 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -201,8 +201,7 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); -int snd_soc_info_bool_ext(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo); +#define snd_soc_info_bool_ext snd_ctl_boolean_mono int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, diff --git a/include/sound/tea575x-tuner.h b/include/sound/tea575x-tuner.h index b5067d3c2387..e8eeb3a1ed29 100644 --- a/include/sound/tea575x-tuner.h +++ b/include/sound/tea575x-tuner.h @@ -4,7 +4,7 @@ /* * ALSA driver for TEA5757/5759 Philips AM/FM tuner chips * - * Copyright (c) 2004 Jaroslav Kysela <perex@suse.cz> + * Copyright (c) 2004 Jaroslav Kysela <perex@perex.cz> * * 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 diff --git a/include/sound/timer.h b/include/sound/timer.h index d42c083db1da..7990469a44ce 100644 --- a/include/sound/timer.h +++ b/include/sound/timer.h @@ -3,7 +3,7 @@ /* * Timer abstract layer - * Copyright (c) by Jaroslav Kysela <perex@suse.cz>, + * Copyright (c) by Jaroslav Kysela <perex@perex.cz>, * Abramo Bagnara <abramo@alsa-project.org> * * diff --git a/include/sound/tlv.h b/include/sound/tlv.h index d93a96b91875..d136ea2181ed 100644 --- a/include/sound/tlv.h +++ b/include/sound/tlv.h @@ -3,7 +3,7 @@ /* * Advanced Linux Sound Architecture - ALSA - Driver - * Copyright (c) 2006 by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) 2006 by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/include/sound/version.h b/include/sound/version.h index 6bbcfefd2c38..8d4a8dd89237 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ /* include/version.h. Generated by alsa/ksync script. */ -#define CONFIG_SND_VERSION "1.0.14" -#define CONFIG_SND_DATE " (Fri Jul 20 09:12:58 2007 UTC)" +#define CONFIG_SND_VERSION "1.0.15" +#define CONFIG_SND_DATE " (Tue Oct 16 14:57:44 2007 UTC)" diff --git a/include/sound/ymfpci.h b/include/sound/ymfpci.h index 203d2b45b788..05ead6698434 100644 --- a/include/sound/ymfpci.h +++ b/include/sound/ymfpci.h @@ -2,7 +2,7 @@ #define __SOUND_YMFPCI_H /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Definitions for Yahama YMF724/740/744/754 chips * * diff --git a/sound/Kconfig b/sound/Kconfig index e48b9b37d228..b2a2db47aff5 100644 --- a/sound/Kconfig +++ b/sound/Kconfig @@ -63,6 +63,10 @@ source "sound/aoa/Kconfig" source "sound/arm/Kconfig" +if SPI +source "sound/spi/Kconfig" +endif + source "sound/mips/Kconfig" source "sound/sh/Kconfig" diff --git a/sound/Makefile b/sound/Makefile index 3ead922bd9c6..c76d70716fa5 100644 --- a/sound/Makefile +++ b/sound/Makefile @@ -5,7 +5,8 @@ obj-$(CONFIG_SOUND) += soundcore.o obj-$(CONFIG_SOUND_PRIME) += sound_firmware.o obj-$(CONFIG_SOUND_PRIME) += oss/ obj-$(CONFIG_DMASOUND) += oss/ -obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ sh/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ soc/ +obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ sh/ synth/ usb/ \ + sparc/ spi/ parisc/ pcmcia/ mips/ soc/ obj-$(CONFIG_SND_AOA) += aoa/ # This one must be compilable even if sound is configured out diff --git a/sound/aoa/codecs/snd-aoa-codec-onyx.c b/sound/aoa/codecs/snd-aoa-codec-onyx.c index 028852374f21..71e3f9360658 100644 --- a/sound/aoa/codecs/snd-aoa-codec-onyx.c +++ b/sound/aoa/codecs/snd-aoa-codec-onyx.c @@ -297,15 +297,7 @@ static struct snd_kcontrol_new capture_source_control = { .put = onyx_snd_capture_source_put, }; -static int onyx_snd_mute_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define onyx_snd_mute_info snd_ctl_boolean_stereo_info static int onyx_snd_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -359,15 +351,7 @@ static struct snd_kcontrol_new mute_control = { }; -static int onyx_snd_single_bit_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define onyx_snd_single_bit_info snd_ctl_boolean_mono_info #define FLAG_POLARITY_INVERT 1 #define FLAG_SPDIFLOCK 2 diff --git a/sound/aoa/codecs/snd-aoa-codec-tas.c b/sound/aoa/codecs/snd-aoa-codec-tas.c index 2f771f57c76f..70c341684794 100644 --- a/sound/aoa/codecs/snd-aoa-codec-tas.c +++ b/sound/aoa/codecs/snd-aoa-codec-tas.c @@ -272,15 +272,7 @@ static struct snd_kcontrol_new volume_control = { .put = tas_snd_vol_put, }; -static int tas_snd_mute_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define tas_snd_mute_info snd_ctl_boolean_stereo_info static int tas_snd_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -431,15 +423,7 @@ static struct snd_kcontrol_new drc_range_control = { .put = tas_snd_drc_range_put, }; -static int tas_snd_drc_switch_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define tas_snd_drc_switch_info snd_ctl_boolean_mono_info static int tas_snd_drc_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -743,6 +727,7 @@ static int tas_switch_clock(struct codec_info_item *cii, enum clock_switch clock return 0; } +#ifdef CONFIG_PM /* we are controlled via i2c and assume that is always up * If that wasn't the case, we'd have to suspend once * our i2c device is suspended, and then take note of that! */ @@ -768,7 +753,6 @@ static int tas_resume(struct tas *tas) return 0; } -#ifdef CONFIG_PM static int _tas_suspend(struct codec_info_item *cii, pm_message_t state) { return tas_suspend(cii->codec_data); @@ -778,7 +762,10 @@ static int _tas_resume(struct codec_info_item *cii) { return tas_resume(cii->codec_data); } -#endif +#else /* CONFIG_PM */ +#define _tas_suspend NULL +#define _tas_resume NULL +#endif /* CONFIG_PM */ static struct codec_info tas_codec_info = { .transfers = tas_transfers, @@ -791,10 +778,8 @@ static struct codec_info tas_codec_info = { .owner = THIS_MODULE, .usable = tas_usable, .switch_clock = tas_switch_clock, -#ifdef CONFIG_PM .suspend = _tas_suspend, .resume = _tas_resume, -#endif }; static int tas_init_codec(struct aoa_codec *codec) diff --git a/sound/aoa/fabrics/snd-aoa-fabric-layout.c b/sound/aoa/fabrics/snd-aoa-fabric-layout.c index 98806283d1b2..8b2ba99d7f8a 100644 --- a/sound/aoa/fabrics/snd-aoa-fabric-layout.c +++ b/sound/aoa/fabrics/snd-aoa-fabric-layout.c @@ -582,15 +582,7 @@ static int layouts_list_items; * make the fabric handle all the card stuff, etc... */ static struct layout_dev *layout_device; -static int control_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define control_info snd_ctl_boolean_mono_info #define AMP_CONTROL(n, description) \ static int n##_control_get(struct snd_kcontrol *kcontrol, \ diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c index e7ed868fa7c0..81c64b09d359 100644 --- a/sound/arm/sa11xx-uda1341.c +++ b/sound/arm/sa11xx-uda1341.c @@ -79,12 +79,6 @@ #include <asm/mach-types.h> #include <asm/dma.h> -#ifdef CONFIG_H3600_HAL -#include <asm/semaphore.h> -#include <asm/uaccess.h> -#include <asm/arch/h3600_hal.h> -#endif - #include <sound/core.h> #include <sound/pcm.h> #include <sound/initval.h> @@ -100,9 +94,6 @@ * We use DMA stuff from 2.4.18-rmk3-hh24 here to be able to compile this * module for Familiar 0.6.1 */ -#ifdef CONFIG_H3600_HAL -#define HH_VERSION 1 -#endif /* {{{ Type definitions */ @@ -238,11 +229,8 @@ static void sa11xx_uda1341_set_samplerate(struct sa11xx_uda1341 *sa11xx_uda1341, rate = 8000; /* Set the external clock generator */ -#ifdef CONFIG_H3600_HAL - h3600_audio_clock(rate); -#else + sa11xx_uda1341_set_audio_clock(rate); -#endif /* Select the clock divisor */ switch (rate) { @@ -307,13 +295,10 @@ static void sa11xx_uda1341_audio_init(struct sa11xx_uda1341 *sa11xx_uda1341) local_irq_restore(flags); /* Enable the audio power */ -#ifdef CONFIG_H3600_HAL - h3600_audio_power(AUDIO_RATE_DEFAULT); -#else + clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET); set_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON); set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); -#endif /* Wait for the UDA1341 to wake up */ mdelay(1); //FIXME - was removed by Perex - Why? @@ -331,21 +316,13 @@ static void sa11xx_uda1341_audio_init(struct sa11xx_uda1341 *sa11xx_uda1341) /* make the left and right channels unswapped (flip the WS latch) */ Ser4SSDR = 0; -#ifdef CONFIG_H3600_HAL - h3600_audio_mute(0); -#else - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); -#endif + clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); } static void sa11xx_uda1341_audio_shutdown(struct sa11xx_uda1341 *sa11xx_uda1341) { /* mute on */ -#ifdef CONFIG_H3600_HAL - h3600_audio_mute(1); -#else set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); -#endif /* disable the audio power and all signals leading to the audio chip */ l3_close(sa11xx_uda1341->uda1341); @@ -354,13 +331,9 @@ static void sa11xx_uda1341_audio_shutdown(struct sa11xx_uda1341 *sa11xx_uda1341) /* power off and mute off */ /* FIXME - is muting off necesary??? */ -#ifdef CONFIG_H3600_HAL - h3600_audio_power(0); - h3600_audio_mute(0); -#else + clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON); clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); -#endif } /* }}} */ diff --git a/sound/core/Makefile b/sound/core/Makefile index 5a01c76d02e8..267039a97bd5 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -1,20 +1,17 @@ # # Makefile for ALSA -# Copyright (c) 1999,2001 by Jaroslav Kysela <perex@suse.cz> +# Copyright (c) 1999,2001 by Jaroslav Kysela <perex@perex.cz> # -snd-objs := sound.o init.o memory.o info.o control.o misc.o device.o -ifeq ($(CONFIG_ISA_DMA_API),y) -snd-objs += isadma.o -endif -ifeq ($(CONFIG_SND_OSSEMUL),y) -snd-objs += sound_oss.o info_oss.o -endif +snd-y := sound.o init.o memory.o info.o control.o misc.o device.o +snd-$(CONFIG_ISA_DMA_API) += isadma.o +snd-$(CONFIG_SND_OSSEMUL) += sound_oss.o info_oss.o snd-pcm-objs := pcm.o pcm_native.o pcm_lib.o pcm_timer.o pcm_misc.o \ pcm_memory.o -snd-page-alloc-objs := memalloc.o sgbuf.o +snd-page-alloc-y := memalloc.o +snd-page-alloc-$(CONFIG_HAS_DMA) += sgbuf.o snd-rawmidi-objs := rawmidi.o snd-timer-objs := timer.o diff --git a/sound/core/control.c b/sound/core/control.c index 1f1ab9c1b668..4c3aa8e10378 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1,6 +1,6 @@ /* * Routines for driver control interface - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -716,8 +716,6 @@ int snd_ctl_elem_read(struct snd_card *card, struct snd_ctl_elem_value *control) return result; } -EXPORT_SYMBOL(snd_ctl_elem_read); - static int snd_ctl_elem_read_user(struct snd_card *card, struct snd_ctl_elem_value __user *_control) { @@ -781,8 +779,6 @@ int snd_ctl_elem_write(struct snd_card *card, struct snd_ctl_file *file, return result; } -EXPORT_SYMBOL(snd_ctl_elem_write); - static int snd_ctl_elem_write_user(struct snd_ctl_file *file, struct snd_ctl_elem_value __user *_control) { @@ -1486,3 +1482,30 @@ int snd_ctl_create(struct snd_card *card) snd_assert(card != NULL, return -ENXIO); return snd_device_new(card, SNDRV_DEV_CONTROL, card, &ops); } + +/* + * Frequently used control callbacks + */ +int snd_ctl_boolean_mono_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +EXPORT_SYMBOL(snd_ctl_boolean_mono_info); + +int snd_ctl_boolean_stereo_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +EXPORT_SYMBOL(snd_ctl_boolean_stereo_info); diff --git a/sound/core/device.c b/sound/core/device.c index 5858b02b0b1d..ea1a0621eefb 100644 --- a/sound/core/device.c +++ b/sound/core/device.c @@ -1,6 +1,6 @@ /* * Device management routines - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index 51ad95b7c894..bfd9d182b8a3 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -1,6 +1,6 @@ /* * Hardware dependent layer - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -31,7 +31,7 @@ #include <sound/hwdep.h> #include <sound/info.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Hardware dependent layer"); MODULE_LICENSE("GPL"); diff --git a/sound/core/info.c b/sound/core/info.c index bf6dbf99528b..1ffd29bb4cd0 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -1,6 +1,6 @@ /* * Information interface for ALSA driver - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/sound/core/info_oss.c b/sound/core/info_oss.c index a444bfe2cf74..435c9399f7a9 100644 --- a/sound/core/info_oss.c +++ b/sound/core/info_oss.c @@ -1,6 +1,6 @@ /* * Information interface for ALSA driver - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/sound/core/init.c b/sound/core/init.c index f2fe35737186..2cb7099eb1e1 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -1,6 +1,6 @@ /* * Initialization routines - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/sound/core/isadma.c b/sound/core/isadma.c index d52398727f0a..eb173cef4f05 100644 --- a/sound/core/isadma.c +++ b/sound/core/isadma.c @@ -1,6 +1,6 @@ /* * ISA DMA support functions - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 9b5656d8bcca..9b4992eab479 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Takashi Iwai <tiwai@suse.de> * * Generic memory allocators @@ -38,7 +38,7 @@ #endif -MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>, Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>, Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Memory allocator for ALSA system."); MODULE_LICENSE("GPL"); @@ -206,6 +206,7 @@ void snd_free_pages(void *ptr, size_t size) * */ +#ifdef CONFIG_HAS_DMA /* allocate the coherent DMA pages */ static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *dma) { @@ -239,6 +240,7 @@ static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, dec_snd_pages(pg); dma_free_coherent(dev, PAGE_SIZE << pg, ptr, dma); } +#endif /* CONFIG_HAS_DMA */ #ifdef CONFIG_SBUS @@ -312,12 +314,14 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, dmab->area = snd_malloc_sbus_pages(device, size, &dmab->addr); break; #endif +#ifdef CONFIG_HAS_DMA case SNDRV_DMA_TYPE_DEV: dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr); break; case SNDRV_DMA_TYPE_DEV_SG: snd_malloc_sgbuf_pages(device, size, dmab, NULL); break; +#endif default: printk(KERN_ERR "snd-malloc: invalid device type %d\n", type); dmab->area = NULL; @@ -383,12 +387,14 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab) snd_free_sbus_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); break; #endif +#ifdef CONFIG_HAS_DMA case SNDRV_DMA_TYPE_DEV: snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); break; case SNDRV_DMA_TYPE_DEV_SG: snd_free_sgbuf_pages(dmab); break; +#endif default: printk(KERN_ERR "snd-malloc: invalid device type %d\n", dmab->dev.type); } diff --git a/sound/core/memory.c b/sound/core/memory.c index 93537ab7c2ac..25b0f056563e 100644 --- a/sound/core/memory.c +++ b/sound/core/memory.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * Misc memory accessors * diff --git a/sound/core/misc.c b/sound/core/misc.c index f78cd000e88d..6cabab8cc537 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -1,6 +1,6 @@ /* * Misc and compatibility things - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/sound/core/oss/Makefile b/sound/core/oss/Makefile index e6d5a045ba27..10a79453245f 100644 --- a/sound/core/oss/Makefile +++ b/sound/core/oss/Makefile @@ -1,12 +1,13 @@ # # Makefile for ALSA -# Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz> +# Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz> # snd-mixer-oss-objs := mixer_oss.o -snd-pcm-oss-objs := pcm_oss.o pcm_plugin.o \ - io.o copy.o linear.o mulaw.o route.o rate.o +snd-pcm-oss-y := pcm_oss.o +snd-pcm-oss-$(CONFIG_SND_PCM_OSS_PLUGINS) += pcm_plugin.o \ + io.o copy.o linear.o mulaw.o route.o rate.o obj-$(CONFIG_SND_MIXER_OSS) += snd-mixer-oss.o obj-$(CONFIG_SND_PCM_OSS) += snd-pcm-oss.o diff --git a/sound/core/oss/copy.c b/sound/core/oss/copy.c index 6658facc5cda..d6a04c2d5a75 100644 --- a/sound/core/oss/copy.c +++ b/sound/core/oss/copy.c @@ -20,9 +20,6 @@ */ #include <sound/driver.h> - -#ifdef CONFIG_SND_PCM_OSS_PLUGINS - #include <linux/time.h> #include <sound/core.h> #include <sound/pcm.h> @@ -88,5 +85,3 @@ int snd_pcm_plugin_build_copy(struct snd_pcm_substream *plug, *r_plugin = plugin; return 0; } - -#endif diff --git a/sound/core/oss/io.c b/sound/core/oss/io.c index b6e7ce30e5a3..3ece39fc48db 100644 --- a/sound/core/oss/io.c +++ b/sound/core/oss/io.c @@ -1,6 +1,6 @@ /* * PCM I/O Plug-In Interface - * Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz> * * * This library is free software; you can redistribute it and/or modify @@ -20,9 +20,6 @@ */ #include <sound/driver.h> - -#ifdef CONFIG_SND_PCM_OSS_PLUGINS - #include <linux/time.h> #include <sound/core.h> #include <sound/pcm.h> @@ -135,5 +132,3 @@ int snd_pcm_plugin_build_io(struct snd_pcm_substream *plug, *r_plugin = plugin; return 0; } - -#endif diff --git a/sound/core/oss/linear.c b/sound/core/oss/linear.c index 5b1bcdc64779..06f96a3e86f6 100644 --- a/sound/core/oss/linear.c +++ b/sound/core/oss/linear.c @@ -1,6 +1,6 @@ /* * Linear conversion Plug-In - * Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz>, + * Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>, * Abramo Bagnara <abramo@alsa-project.org> * * @@ -21,9 +21,6 @@ */ #include <sound/driver.h> - -#ifdef CONFIG_SND_PCM_OSS_PLUGINS - #include <linux/time.h> #include <sound/core.h> #include <sound/pcm.h> @@ -34,19 +31,34 @@ */ struct linear_priv { - int conv; + int cvt_endian; /* need endian conversion? */ + unsigned int src_ofs; /* byte offset in source format */ + unsigned int dst_ofs; /* byte soffset in destination format */ + unsigned int copy_ofs; /* byte offset in temporary u32 data */ + unsigned int dst_bytes; /* byte size of destination format */ + unsigned int copy_bytes; /* bytes to copy per conversion */ + unsigned int flip; /* MSB flip for signeness, done after endian conv */ }; +static inline void do_convert(struct linear_priv *data, + unsigned char *dst, unsigned char *src) +{ + unsigned int tmp = 0; + unsigned char *p = (unsigned char *)&tmp; + + memcpy(p + data->copy_ofs, src + data->src_ofs, data->copy_bytes); + if (data->cvt_endian) + tmp = swab32(tmp); + tmp ^= data->flip; + memcpy(dst, p + data->dst_ofs, data->dst_bytes); +} + static void convert(struct snd_pcm_plugin *plugin, const struct snd_pcm_plugin_channel *src_channels, struct snd_pcm_plugin_channel *dst_channels, snd_pcm_uframes_t frames) { -#define CONV_LABELS -#include "plugin_ops.h" -#undef CONV_LABELS struct linear_priv *data = (struct linear_priv *)plugin->extra_data; - void *conv = conv_labels[data->conv]; int channel; int nchannels = plugin->src_format.channels; for (channel = 0; channel < nchannels; ++channel) { @@ -67,11 +79,7 @@ static void convert(struct snd_pcm_plugin *plugin, dst_step = dst_channels[channel].area.step / 8; frames1 = frames; while (frames1-- > 0) { - goto *conv; -#define CONV_END after -#include "plugin_ops.h" -#undef CONV_END - after: + do_convert(data, dst, src); src += src_step; dst += dst_step; } @@ -106,29 +114,36 @@ static snd_pcm_sframes_t linear_transfer(struct snd_pcm_plugin *plugin, return frames; } -static int conv_index(int src_format, int dst_format) +static void init_data(struct linear_priv *data, int src_format, int dst_format) { - int src_endian, dst_endian, sign, src_width, dst_width; - - sign = (snd_pcm_format_signed(src_format) != - snd_pcm_format_signed(dst_format)); -#ifdef SNDRV_LITTLE_ENDIAN - src_endian = snd_pcm_format_big_endian(src_format); - dst_endian = snd_pcm_format_big_endian(dst_format); -#else - src_endian = snd_pcm_format_little_endian(src_format); - dst_endian = snd_pcm_format_little_endian(dst_format); -#endif - - if (src_endian < 0) - src_endian = 0; - if (dst_endian < 0) - dst_endian = 0; - - src_width = snd_pcm_format_width(src_format) / 8 - 1; - dst_width = snd_pcm_format_width(dst_format) / 8 - 1; - - return src_width * 32 + src_endian * 16 + sign * 8 + dst_width * 2 + dst_endian; + int src_le, dst_le, src_bytes, dst_bytes; + + src_bytes = snd_pcm_format_width(src_format) / 8; + dst_bytes = snd_pcm_format_width(dst_format) / 8; + src_le = snd_pcm_format_little_endian(src_format) > 0; + dst_le = snd_pcm_format_little_endian(dst_format) > 0; + + data->dst_bytes = dst_bytes; + data->cvt_endian = src_le != dst_le; + data->copy_bytes = src_bytes < dst_bytes ? src_bytes : dst_bytes; + if (src_le) { + data->copy_ofs = 4 - data->copy_bytes; + data->src_ofs = src_bytes - data->copy_bytes; + } else + data->src_ofs = snd_pcm_format_physical_width(src_format) / 8 - + src_bytes; + if (dst_le) + data->dst_ofs = 4 - data->dst_bytes; + else + data->dst_ofs = snd_pcm_format_physical_width(dst_format) / 8 - + dst_bytes; + if (snd_pcm_format_signed(src_format) != + snd_pcm_format_signed(dst_format)) { + if (dst_le) + data->flip = cpu_to_le32(0x80000000); + else + data->flip = cpu_to_be32(0x80000000); + } } int snd_pcm_plugin_build_linear(struct snd_pcm_substream *plug, @@ -154,10 +169,8 @@ int snd_pcm_plugin_build_linear(struct snd_pcm_substream *plug, if (err < 0) return err; data = (struct linear_priv *)plugin->extra_data; - data->conv = conv_index(src_format->format, dst_format->format); + init_data(data, src_format->format, dst_format->format); plugin->transfer = linear_transfer; *r_plugin = plugin; return 0; } - -#endif diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index fccad8f0a6bb..3ace4a5680ba 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -1,6 +1,6 @@ /* * OSS emulation layer for the mixer interface - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -33,7 +33,7 @@ #define OSS_ALSAEMULVER _SIOR ('M', 249, int) -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Mixer OSS emulation for ALSA."); MODULE_LICENSE("GPL"); MODULE_ALIAS_SNDRV_MINOR(SNDRV_MINOR_OSS_MIXER); diff --git a/sound/core/oss/mulaw.c b/sound/core/oss/mulaw.c index 2eb18807e6d0..848db82529ed 100644 --- a/sound/core/oss/mulaw.c +++ b/sound/core/oss/mulaw.c @@ -1,6 +1,6 @@ /* * Mu-Law conversion Plug-In Interface - * Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz> * Uros Bizjak <uros@kss-loka.si> * * Based on reference implementation by Sun Microsystems, Inc. @@ -22,9 +22,6 @@ */ #include <sound/driver.h> - -#ifdef CONFIG_SND_PCM_OSS_PLUGINS - #include <linux/time.h> #include <sound/core.h> #include <sound/pcm.h> @@ -149,19 +146,32 @@ typedef void (*mulaw_f)(struct snd_pcm_plugin *plugin, struct mulaw_priv { mulaw_f func; - int conv; + int cvt_endian; /* need endian conversion? */ + unsigned int native_ofs; /* byte offset in native format */ + unsigned int copy_ofs; /* byte offset in s16 format */ + unsigned int native_bytes; /* byte size of the native format */ + unsigned int copy_bytes; /* bytes to copy per conversion */ + u16 flip; /* MSB flip for signedness, done after endian conversion */ }; +static inline void cvt_s16_to_native(struct mulaw_priv *data, + unsigned char *dst, u16 sample) +{ + sample ^= data->flip; + if (data->cvt_endian) + sample = swab16(sample); + if (data->native_bytes > data->copy_bytes) + memset(dst, 0, data->native_bytes); + memcpy(dst + data->native_ofs, (char *)&sample + data->copy_ofs, + data->copy_bytes); +} + static void mulaw_decode(struct snd_pcm_plugin *plugin, const struct snd_pcm_plugin_channel *src_channels, struct snd_pcm_plugin_channel *dst_channels, snd_pcm_uframes_t frames) { -#define PUT_S16_LABELS -#include "plugin_ops.h" -#undef PUT_S16_LABELS struct mulaw_priv *data = (struct mulaw_priv *)plugin->extra_data; - void *put = put_s16_labels[data->conv]; int channel; int nchannels = plugin->src_format.channels; for (channel = 0; channel < nchannels; ++channel) { @@ -183,30 +193,33 @@ static void mulaw_decode(struct snd_pcm_plugin *plugin, frames1 = frames; while (frames1-- > 0) { signed short sample = ulaw2linear(*src); - goto *put; -#define PUT_S16_END after -#include "plugin_ops.h" -#undef PUT_S16_END - after: + cvt_s16_to_native(data, dst, sample); src += src_step; dst += dst_step; } } } +static inline signed short cvt_native_to_s16(struct mulaw_priv *data, + unsigned char *src) +{ + u16 sample = 0; + memcpy((char *)&sample + data->copy_ofs, src + data->native_ofs, + data->copy_bytes); + if (data->cvt_endian) + sample = swab16(sample); + sample ^= data->flip; + return (signed short)sample; +} + static void mulaw_encode(struct snd_pcm_plugin *plugin, const struct snd_pcm_plugin_channel *src_channels, struct snd_pcm_plugin_channel *dst_channels, snd_pcm_uframes_t frames) { -#define GET_S16_LABELS -#include "plugin_ops.h" -#undef GET_S16_LABELS struct mulaw_priv *data = (struct mulaw_priv *)plugin->extra_data; - void *get = get_s16_labels[data->conv]; int channel; int nchannels = plugin->src_format.channels; - signed short sample = 0; for (channel = 0; channel < nchannels; ++channel) { char *src; char *dst; @@ -225,11 +238,7 @@ static void mulaw_encode(struct snd_pcm_plugin *plugin, dst_step = dst_channels[channel].area.step / 8; frames1 = frames; while (frames1-- > 0) { - goto *get; -#define GET_S16_END after -#include "plugin_ops.h" -#undef GET_S16_END - after: + signed short sample = cvt_native_to_s16(data, src); *dst = linear2ulaw(sample); src += src_step; dst += dst_step; @@ -265,23 +274,25 @@ static snd_pcm_sframes_t mulaw_transfer(struct snd_pcm_plugin *plugin, return frames; } -static int getput_index(int format) +static void init_data(struct mulaw_priv *data, int format) { - int sign, width, endian; - sign = !snd_pcm_format_signed(format); - width = snd_pcm_format_width(format) / 8 - 1; - if (width < 0 || width > 3) { - snd_printk(KERN_ERR "snd-pcm-oss: invalid format %d\n", format); - width = 0; - } #ifdef SNDRV_LITTLE_ENDIAN - endian = snd_pcm_format_big_endian(format); + data->cvt_endian = snd_pcm_format_big_endian(format) > 0; #else - endian = snd_pcm_format_little_endian(format); + data->cvt_endian = snd_pcm_format_little_endian(format) > 0; #endif - if (endian < 0) - endian = 0; - return width * 4 + endian * 2 + sign; + if (!snd_pcm_format_signed(format)) + data->flip = 0x8000; + data->native_bytes = snd_pcm_format_physical_width(format) / 8; + data->copy_bytes = data->native_bytes < 2 ? 1 : 2; + if (snd_pcm_format_little_endian(format)) { + data->native_ofs = data->native_bytes - data->copy_bytes; + data->copy_ofs = 2 - data->copy_bytes; + } else { + /* S24 in 4bytes need an 1 byte offset */ + data->native_ofs = data->native_bytes - + snd_pcm_format_width(format) / 8; + } } int snd_pcm_plugin_build_mulaw(struct snd_pcm_substream *plug, @@ -322,11 +333,8 @@ int snd_pcm_plugin_build_mulaw(struct snd_pcm_substream *plug, return err; data = (struct mulaw_priv *)plugin->extra_data; data->func = func; - data->conv = getput_index(format->format); - snd_assert(data->conv >= 0 && data->conv < 4*2*2, return -EINVAL); + init_data(data, format->format); plugin->transfer = mulaw_transfer; *r_plugin = plugin; return 0; } - -#endif diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index fc11572c48cf..d0c4ceb9f0b4 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1,6 +1,6 @@ /* * Digital Audio (PCM) abstract layer / OSS compatible - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -48,7 +48,7 @@ static int dsp_map[SNDRV_CARDS]; static int adsp_map[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = 1}; static int nonblock_open = 1; -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>, Abramo Bagnara <abramo@alsa-project.org>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, Abramo Bagnara <abramo@alsa-project.org>"); MODULE_DESCRIPTION("PCM OSS emulation for ALSA."); MODULE_LICENSE("GPL"); module_param_array(dsp_map, int, NULL, 0444); @@ -633,6 +633,22 @@ static long snd_pcm_alsa_frames(struct snd_pcm_substream *substream, long bytes) return bytes_to_frames(runtime, (buffer_size * bytes) / runtime->oss.buffer_bytes); } +/* define extended formats in the recent OSS versions (if any) */ +/* linear formats */ +#define AFMT_S32_LE 0x00001000 +#define AFMT_S32_BE 0x00002000 +#define AFMT_S24_LE 0x00008000 +#define AFMT_S24_BE 0x00010000 +#define AFMT_S24_PACKED 0x00040000 + +/* other supported formats */ +#define AFMT_FLOAT 0x00004000 +#define AFMT_SPDIF_RAW 0x00020000 + +/* unsupported formats */ +#define AFMT_AC3 0x00000400 +#define AFMT_VORBIS 0x00000800 + static int snd_pcm_oss_format_from(int format) { switch (format) { @@ -646,6 +662,13 @@ static int snd_pcm_oss_format_from(int format) case AFMT_U16_LE: return SNDRV_PCM_FORMAT_U16_LE; case AFMT_U16_BE: return SNDRV_PCM_FORMAT_U16_BE; case AFMT_MPEG: return SNDRV_PCM_FORMAT_MPEG; + case AFMT_S32_LE: return SNDRV_PCM_FORMAT_S32_LE; + case AFMT_S32_BE: return SNDRV_PCM_FORMAT_S32_BE; + case AFMT_S24_LE: return SNDRV_PCM_FORMAT_S24_LE; + case AFMT_S24_BE: return SNDRV_PCM_FORMAT_S24_BE; + case AFMT_S24_PACKED: return SNDRV_PCM_FORMAT_S24_3LE; + case AFMT_FLOAT: return SNDRV_PCM_FORMAT_FLOAT; + case AFMT_SPDIF_RAW: return SNDRV_PCM_FORMAT_IEC958_SUBFRAME; default: return SNDRV_PCM_FORMAT_U8; } } @@ -663,6 +686,13 @@ static int snd_pcm_oss_format_to(int format) case SNDRV_PCM_FORMAT_U16_LE: return AFMT_U16_LE; case SNDRV_PCM_FORMAT_U16_BE: return AFMT_U16_BE; case SNDRV_PCM_FORMAT_MPEG: return AFMT_MPEG; + case SNDRV_PCM_FORMAT_S32_LE: return AFMT_S32_LE; + case SNDRV_PCM_FORMAT_S32_BE: return AFMT_S32_BE; + case SNDRV_PCM_FORMAT_S24_LE: return AFMT_S24_LE; + case SNDRV_PCM_FORMAT_S24_BE: return AFMT_S24_BE; + case SNDRV_PCM_FORMAT_S24_3LE: return AFMT_S24_PACKED; + case SNDRV_PCM_FORMAT_FLOAT: return AFMT_FLOAT; + case SNDRV_PCM_FORMAT_IEC958_SUBFRAME: return AFMT_SPDIF_RAW; default: return -EINVAL; } } @@ -1725,7 +1755,10 @@ static int snd_pcm_oss_get_formats(struct snd_pcm_oss_file *pcm_oss_file) return AFMT_MU_LAW | AFMT_U8 | AFMT_S16_LE | AFMT_S16_BE | AFMT_S8 | AFMT_U16_LE | - AFMT_U16_BE; + AFMT_U16_BE | + AFMT_S32_LE | AFMT_S32_BE | + AFMT_S24_LE | AFMT_S24_LE | + AFMT_S24_PACKED; params = kmalloc(sizeof(*params), GFP_KERNEL); if (!params) return -ENOMEM; diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 0e67dd280a5d..14095a927a1b 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -1,6 +1,6 @@ /* * PCM Plug-In shared (kernel/library) code - * Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz> * Copyright (c) 2000 by Abramo Bagnara <abramo@alsa-project.org> * * @@ -25,9 +25,6 @@ #endif #include <sound/driver.h> - -#ifdef CONFIG_SND_PCM_OSS_PLUGINS - #include <linux/slab.h> #include <linux/time.h> #include <linux/vmalloc.h> @@ -267,6 +264,8 @@ static int snd_pcm_plug_formats(struct snd_mask *mask, int format) SNDRV_PCM_FMTBIT_U16_BE | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_BE | SNDRV_PCM_FMTBIT_S24_BE | + SNDRV_PCM_FMTBIT_U24_3LE | SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_U24_3BE | SNDRV_PCM_FMTBIT_S24_3BE | SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_BE | SNDRV_PCM_FMTBIT_S32_BE); snd_mask_set(&formats, SNDRV_PCM_FORMAT_MU_LAW); @@ -283,6 +282,10 @@ static int preferred_formats[] = { SNDRV_PCM_FORMAT_S16_BE, SNDRV_PCM_FORMAT_U16_LE, SNDRV_PCM_FORMAT_U16_BE, + SNDRV_PCM_FORMAT_S24_3LE, + SNDRV_PCM_FORMAT_S24_3BE, + SNDRV_PCM_FORMAT_U24_3LE, + SNDRV_PCM_FORMAT_U24_3BE, SNDRV_PCM_FORMAT_S24_LE, SNDRV_PCM_FORMAT_S24_BE, SNDRV_PCM_FORMAT_U24_LE, @@ -297,41 +300,37 @@ static int preferred_formats[] = { int snd_pcm_plug_slave_format(int format, struct snd_mask *format_mask) { + int i; + if (snd_mask_test(format_mask, format)) return format; if (! snd_pcm_plug_formats(format_mask, format)) return -EINVAL; if (snd_pcm_format_linear(format)) { - int width = snd_pcm_format_width(format); - int unsignd = snd_pcm_format_unsigned(format); - int big = snd_pcm_format_big_endian(format); - int format1; - int wid, width1=width; - int dwidth1 = 8; - for (wid = 0; wid < 4; ++wid) { - int end, big1 = big; - for (end = 0; end < 2; ++end) { - int sgn, unsignd1 = unsignd; - for (sgn = 0; sgn < 2; ++sgn) { - format1 = snd_pcm_build_linear_format(width1, unsignd1, big1); - if (format1 >= 0 && - snd_mask_test(format_mask, format1)) - goto _found; - unsignd1 = !unsignd1; - } - big1 = !big1; - } - if (width1 == 32) { - dwidth1 = -dwidth1; - width1 = width; + unsigned int width = snd_pcm_format_width(format); + int unsignd = snd_pcm_format_unsigned(format) > 0; + int big = snd_pcm_format_big_endian(format) > 0; + unsigned int badness, best = -1; + int best_format = -1; + for (i = 0; i < ARRAY_SIZE(preferred_formats); i++) { + int f = preferred_formats[i]; + unsigned int w; + if (!snd_mask_test(format_mask, f)) + continue; + w = snd_pcm_format_width(f); + if (w >= width) + badness = w - width; + else + badness = width - w + 32; + badness += snd_pcm_format_unsigned(f) != unsignd; + badness += snd_pcm_format_big_endian(f) != big; + if (badness < best) { + best_format = f; + best = badness; } - width1 += dwidth1; } - return -EINVAL; - _found: - return format1; + return best_format >= 0 ? best_format : -EINVAL; } else { - unsigned int i; switch (format) { case SNDRV_PCM_FORMAT_MU_LAW: for (i = 0; i < ARRAY_SIZE(preferred_formats); ++i) { @@ -740,5 +739,3 @@ int snd_pcm_area_copy(const struct snd_pcm_channel_area *src_area, size_t src_of } return 0; } - -#endif diff --git a/sound/core/oss/pcm_plugin.h b/sound/core/oss/pcm_plugin.h index 3be91b3d5377..ca2f4c39be46 100644 --- a/sound/core/oss/pcm_plugin.h +++ b/sound/core/oss/pcm_plugin.h @@ -3,7 +3,7 @@ /* * Digital Audio (Plugin interface) abstract layer - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/sound/core/oss/plugin_ops.h b/sound/core/oss/plugin_ops.h deleted file mode 100644 index 1f5bde4631f1..000000000000 --- a/sound/core/oss/plugin_ops.h +++ /dev/null @@ -1,370 +0,0 @@ -/* - * Plugin sample operators with fast switch - * Copyright (c) 2000 by Jaroslav Kysela <perex@suse.cz> - * - * - * This library is free software; you can redistribute it and/or modify - * it under the terms of the GNU Library 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 Library General Public License for more details. - * - * You should have received a copy of the GNU Library General Public - * License along with this library; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ - - -#define as_u8(ptr) (*(u_int8_t*)(ptr)) -#define as_u16(ptr) (*(u_int16_t*)(ptr)) -#define as_u32(ptr) (*(u_int32_t*)(ptr)) -#define as_u64(ptr) (*(u_int64_t*)(ptr)) -#define as_s8(ptr) (*(int8_t*)(ptr)) -#define as_s16(ptr) (*(int16_t*)(ptr)) -#define as_s32(ptr) (*(int32_t*)(ptr)) -#define as_s64(ptr) (*(int64_t*)(ptr)) - -#ifdef COPY_LABELS -static void *copy_labels[4] = { - &©_8, - &©_16, - &©_32, - &©_64 -}; -#endif - -#ifdef COPY_END -while(0) { -copy_8: as_s8(dst) = as_s8(src); goto COPY_END; -copy_16: as_s16(dst) = as_s16(src); goto COPY_END; -copy_32: as_s32(dst) = as_s32(src); goto COPY_END; -copy_64: as_s64(dst) = as_s64(src); goto COPY_END; -} -#endif - -#ifdef CONV_LABELS -/* src_wid src_endswap sign_toggle dst_wid dst_endswap */ -static void *conv_labels[4 * 2 * 2 * 4 * 2] = { - &&conv_xxx1_xxx1, /* 8h -> 8h */ - &&conv_xxx1_xxx1, /* 8h -> 8s */ - &&conv_xxx1_xx10, /* 8h -> 16h */ - &&conv_xxx1_xx01, /* 8h -> 16s */ - &&conv_xxx1_x100, /* 8h -> 24h */ - &&conv_xxx1_001x, /* 8h -> 24s */ - &&conv_xxx1_1000, /* 8h -> 32h */ - &&conv_xxx1_0001, /* 8h -> 32s */ - &&conv_xxx1_xxx9, /* 8h ^> 8h */ - &&conv_xxx1_xxx9, /* 8h ^> 8s */ - &&conv_xxx1_xx90, /* 8h ^> 16h */ - &&conv_xxx1_xx09, /* 8h ^> 16s */ - &&conv_xxx1_x900, /* 8h ^> 24h */ - &&conv_xxx1_009x, /* 8h ^> 24s */ - &&conv_xxx1_9000, /* 8h ^> 32h */ - &&conv_xxx1_0009, /* 8h ^> 32s */ - &&conv_xxx1_xxx1, /* 8s -> 8h */ - &&conv_xxx1_xxx1, /* 8s -> 8s */ - &&conv_xxx1_xx10, /* 8s -> 16h */ - &&conv_xxx1_xx01, /* 8s -> 16s */ - &&conv_xxx1_x100, /* 8s -> 24h */ - &&conv_xxx1_001x, /* 8s -> 24s */ - &&conv_xxx1_1000, /* 8s -> 32h */ - &&conv_xxx1_0001, /* 8s -> 32s */ - &&conv_xxx1_xxx9, /* 8s ^> 8h */ - &&conv_xxx1_xxx9, /* 8s ^> 8s */ - &&conv_xxx1_xx90, /* 8s ^> 16h */ - &&conv_xxx1_xx09, /* 8s ^> 16s */ - &&conv_xxx1_x900, /* 8s ^> 24h */ - &&conv_xxx1_009x, /* 8s ^> 24s */ - &&conv_xxx1_9000, /* 8s ^> 32h */ - &&conv_xxx1_0009, /* 8s ^> 32s */ - &&conv_xx12_xxx1, /* 16h -> 8h */ - &&conv_xx12_xxx1, /* 16h -> 8s */ - &&conv_xx12_xx12, /* 16h -> 16h */ - &&conv_xx12_xx21, /* 16h -> 16s */ - &&conv_xx12_x120, /* 16h -> 24h */ - &&conv_xx12_021x, /* 16h -> 24s */ - &&conv_xx12_1200, /* 16h -> 32h */ - &&conv_xx12_0021, /* 16h -> 32s */ - &&conv_xx12_xxx9, /* 16h ^> 8h */ - &&conv_xx12_xxx9, /* 16h ^> 8s */ - &&conv_xx12_xx92, /* 16h ^> 16h */ - &&conv_xx12_xx29, /* 16h ^> 16s */ - &&conv_xx12_x920, /* 16h ^> 24h */ - &&conv_xx12_029x, /* 16h ^> 24s */ - &&conv_xx12_9200, /* 16h ^> 32h */ - &&conv_xx12_0029, /* 16h ^> 32s */ - &&conv_xx12_xxx2, /* 16s -> 8h */ - &&conv_xx12_xxx2, /* 16s -> 8s */ - &&conv_xx12_xx21, /* 16s -> 16h */ - &&conv_xx12_xx12, /* 16s -> 16s */ - &&conv_xx12_x210, /* 16s -> 24h */ - &&conv_xx12_012x, /* 16s -> 24s */ - &&conv_xx12_2100, /* 16s -> 32h */ - &&conv_xx12_0012, /* 16s -> 32s */ - &&conv_xx12_xxxA, /* 16s ^> 8h */ - &&conv_xx12_xxxA, /* 16s ^> 8s */ - &&conv_xx12_xxA1, /* 16s ^> 16h */ - &&conv_xx12_xx1A, /* 16s ^> 16s */ - &&conv_xx12_xA10, /* 16s ^> 24h */ - &&conv_xx12_01Ax, /* 16s ^> 24s */ - &&conv_xx12_A100, /* 16s ^> 32h */ - &&conv_xx12_001A, /* 16s ^> 32s */ - &&conv_x123_xxx1, /* 24h -> 8h */ - &&conv_x123_xxx1, /* 24h -> 8s */ - &&conv_x123_xx12, /* 24h -> 16h */ - &&conv_x123_xx21, /* 24h -> 16s */ - &&conv_x123_x123, /* 24h -> 24h */ - &&conv_x123_321x, /* 24h -> 24s */ - &&conv_x123_1230, /* 24h -> 32h */ - &&conv_x123_0321, /* 24h -> 32s */ - &&conv_x123_xxx9, /* 24h ^> 8h */ - &&conv_x123_xxx9, /* 24h ^> 8s */ - &&conv_x123_xx92, /* 24h ^> 16h */ - &&conv_x123_xx29, /* 24h ^> 16s */ - &&conv_x123_x923, /* 24h ^> 24h */ - &&conv_x123_329x, /* 24h ^> 24s */ - &&conv_x123_9230, /* 24h ^> 32h */ - &&conv_x123_0329, /* 24h ^> 32s */ - &&conv_123x_xxx3, /* 24s -> 8h */ - &&conv_123x_xxx3, /* 24s -> 8s */ - &&conv_123x_xx32, /* 24s -> 16h */ - &&conv_123x_xx23, /* 24s -> 16s */ - &&conv_123x_x321, /* 24s -> 24h */ - &&conv_123x_123x, /* 24s -> 24s */ - &&conv_123x_3210, /* 24s -> 32h */ - &&conv_123x_0123, /* 24s -> 32s */ - &&conv_123x_xxxB, /* 24s ^> 8h */ - &&conv_123x_xxxB, /* 24s ^> 8s */ - &&conv_123x_xxB2, /* 24s ^> 16h */ - &&conv_123x_xx2B, /* 24s ^> 16s */ - &&conv_123x_xB21, /* 24s ^> 24h */ - &&conv_123x_12Bx, /* 24s ^> 24s */ - &&conv_123x_B210, /* 24s ^> 32h */ - &&conv_123x_012B, /* 24s ^> 32s */ - &&conv_1234_xxx1, /* 32h -> 8h */ - &&conv_1234_xxx1, /* 32h -> 8s */ - &&conv_1234_xx12, /* 32h -> 16h */ - &&conv_1234_xx21, /* 32h -> 16s */ - &&conv_1234_x123, /* 32h -> 24h */ - &&conv_1234_321x, /* 32h -> 24s */ - &&conv_1234_1234, /* 32h -> 32h */ - &&conv_1234_4321, /* 32h -> 32s */ - &&conv_1234_xxx9, /* 32h ^> 8h */ - &&conv_1234_xxx9, /* 32h ^> 8s */ - &&conv_1234_xx92, /* 32h ^> 16h */ - &&conv_1234_xx29, /* 32h ^> 16s */ - &&conv_1234_x923, /* 32h ^> 24h */ - &&conv_1234_329x, /* 32h ^> 24s */ - &&conv_1234_9234, /* 32h ^> 32h */ - &&conv_1234_4329, /* 32h ^> 32s */ - &&conv_1234_xxx4, /* 32s -> 8h */ - &&conv_1234_xxx4, /* 32s -> 8s */ - &&conv_1234_xx43, /* 32s -> 16h */ - &&conv_1234_xx34, /* 32s -> 16s */ - &&conv_1234_x432, /* 32s -> 24h */ - &&conv_1234_234x, /* 32s -> 24s */ - &&conv_1234_4321, /* 32s -> 32h */ - &&conv_1234_1234, /* 32s -> 32s */ - &&conv_1234_xxxC, /* 32s ^> 8h */ - &&conv_1234_xxxC, /* 32s ^> 8s */ - &&conv_1234_xxC3, /* 32s ^> 16h */ - &&conv_1234_xx3C, /* 32s ^> 16s */ - &&conv_1234_xC32, /* 32s ^> 24h */ - &&conv_1234_23Cx, /* 32s ^> 24s */ - &&conv_1234_C321, /* 32s ^> 32h */ - &&conv_1234_123C, /* 32s ^> 32s */ -}; -#endif - -#ifdef CONV_END -while(0) { -conv_xxx1_xxx1: as_u8(dst) = as_u8(src); goto CONV_END; -conv_xxx1_xx10: as_u16(dst) = (u_int16_t)as_u8(src) << 8; goto CONV_END; -conv_xxx1_xx01: as_u16(dst) = (u_int16_t)as_u8(src); goto CONV_END; -conv_xxx1_x100: as_u32(dst) = (u_int32_t)as_u8(src) << 16; goto CONV_END; -conv_xxx1_001x: as_u32(dst) = (u_int32_t)as_u8(src) << 8; goto CONV_END; -conv_xxx1_1000: as_u32(dst) = (u_int32_t)as_u8(src) << 24; goto CONV_END; -conv_xxx1_0001: as_u32(dst) = (u_int32_t)as_u8(src); goto CONV_END; -conv_xxx1_xxx9: as_u8(dst) = as_u8(src) ^ 0x80; goto CONV_END; -conv_xxx1_xx90: as_u16(dst) = (u_int16_t)(as_u8(src) ^ 0x80) << 8; goto CONV_END; -conv_xxx1_xx09: as_u16(dst) = (u_int16_t)(as_u8(src) ^ 0x80); goto CONV_END; -conv_xxx1_x900: as_u32(dst) = (u_int32_t)(as_u8(src) ^ 0x80) << 16; goto CONV_END; -conv_xxx1_009x: as_u32(dst) = (u_int32_t)(as_u8(src) ^ 0x80) << 8; goto CONV_END; -conv_xxx1_9000: as_u32(dst) = (u_int32_t)(as_u8(src) ^ 0x80) << 24; goto CONV_END; -conv_xxx1_0009: as_u32(dst) = (u_int32_t)(as_u8(src) ^ 0x80); goto CONV_END; -conv_xx12_xxx1: as_u8(dst) = as_u16(src) >> 8; goto CONV_END; -conv_xx12_xx12: as_u16(dst) = as_u16(src); goto CONV_END; -conv_xx12_xx21: as_u16(dst) = swab16(as_u16(src)); goto CONV_END; -conv_xx12_x120: as_u32(dst) = (u_int32_t)as_u16(src) << 8; goto CONV_END; -conv_xx12_021x: as_u32(dst) = (u_int32_t)swab16(as_u16(src)) << 8; goto CONV_END; -conv_xx12_1200: as_u32(dst) = (u_int32_t)as_u16(src) << 16; goto CONV_END; -conv_xx12_0021: as_u32(dst) = (u_int32_t)swab16(as_u16(src)); goto CONV_END; -conv_xx12_xxx9: as_u8(dst) = (as_u16(src) >> 8) ^ 0x80; goto CONV_END; -conv_xx12_xx92: as_u16(dst) = as_u16(src) ^ 0x8000; goto CONV_END; -conv_xx12_xx29: as_u16(dst) = swab16(as_u16(src)) ^ 0x80; goto CONV_END; -conv_xx12_x920: as_u32(dst) = (u_int32_t)(as_u16(src) ^ 0x8000) << 8; goto CONV_END; -conv_xx12_029x: as_u32(dst) = (u_int32_t)(swab16(as_u16(src)) ^ 0x80) << 8; goto CONV_END; -conv_xx12_9200: as_u32(dst) = (u_int32_t)(as_u16(src) ^ 0x8000) << 16; goto CONV_END; -conv_xx12_0029: as_u32(dst) = (u_int32_t)(swab16(as_u16(src)) ^ 0x80); goto CONV_END; -conv_xx12_xxx2: as_u8(dst) = as_u16(src) & 0xff; goto CONV_END; -conv_xx12_x210: as_u32(dst) = (u_int32_t)swab16(as_u16(src)) << 8; goto CONV_END; -conv_xx12_012x: as_u32(dst) = (u_int32_t)as_u16(src) << 8; goto CONV_END; -conv_xx12_2100: as_u32(dst) = (u_int32_t)swab16(as_u16(src)) << 16; goto CONV_END; -conv_xx12_0012: as_u32(dst) = (u_int32_t)as_u16(src); goto CONV_END; -conv_xx12_xxxA: as_u8(dst) = (as_u16(src) ^ 0x80) & 0xff; goto CONV_END; -conv_xx12_xxA1: as_u16(dst) = swab16(as_u16(src) ^ 0x80); goto CONV_END; -conv_xx12_xx1A: as_u16(dst) = as_u16(src) ^ 0x80; goto CONV_END; -conv_xx12_xA10: as_u32(dst) = (u_int32_t)swab16(as_u16(src) ^ 0x80) << 8; goto CONV_END; -conv_xx12_01Ax: as_u32(dst) = (u_int32_t)(as_u16(src) ^ 0x80) << 8; goto CONV_END; -conv_xx12_A100: as_u32(dst) = (u_int32_t)swab16(as_u16(src) ^ 0x80) << 16; goto CONV_END; -conv_xx12_001A: as_u32(dst) = (u_int32_t)(as_u16(src) ^ 0x80); goto CONV_END; -conv_x123_xxx1: as_u8(dst) = as_u32(src) >> 16; goto CONV_END; -conv_x123_xx12: as_u16(dst) = as_u32(src) >> 8; goto CONV_END; -conv_x123_xx21: as_u16(dst) = swab16(as_u32(src) >> 8); goto CONV_END; -conv_x123_x123: as_u32(dst) = as_u32(src); goto CONV_END; -conv_x123_321x: as_u32(dst) = swab32(as_u32(src)); goto CONV_END; -conv_x123_1230: as_u32(dst) = as_u32(src) << 8; goto CONV_END; -conv_x123_0321: as_u32(dst) = swab32(as_u32(src)) >> 8; goto CONV_END; -conv_x123_xxx9: as_u8(dst) = (as_u32(src) >> 16) ^ 0x80; goto CONV_END; -conv_x123_xx92: as_u16(dst) = (as_u32(src) >> 8) ^ 0x8000; goto CONV_END; -conv_x123_xx29: as_u16(dst) = swab16(as_u32(src) >> 8) ^ 0x80; goto CONV_END; -conv_x123_x923: as_u32(dst) = as_u32(src) ^ 0x800000; goto CONV_END; -conv_x123_329x: as_u32(dst) = swab32(as_u32(src)) ^ 0x8000; goto CONV_END; -conv_x123_9230: as_u32(dst) = (as_u32(src) ^ 0x800000) << 8; goto CONV_END; -conv_x123_0329: as_u32(dst) = (swab32(as_u32(src)) >> 8) ^ 0x80; goto CONV_END; -conv_123x_xxx3: as_u8(dst) = (as_u32(src) >> 8) & 0xff; goto CONV_END; -conv_123x_xx32: as_u16(dst) = swab16(as_u32(src) >> 8); goto CONV_END; -conv_123x_xx23: as_u16(dst) = (as_u32(src) >> 8) & 0xffff; goto CONV_END; -conv_123x_x321: as_u32(dst) = swab32(as_u32(src)); goto CONV_END; -conv_123x_123x: as_u32(dst) = as_u32(src); goto CONV_END; -conv_123x_3210: as_u32(dst) = swab32(as_u32(src)) << 8; goto CONV_END; -conv_123x_0123: as_u32(dst) = as_u32(src) >> 8; goto CONV_END; -conv_123x_xxxB: as_u8(dst) = ((as_u32(src) >> 8) & 0xff) ^ 0x80; goto CONV_END; -conv_123x_xxB2: as_u16(dst) = swab16((as_u32(src) >> 8) ^ 0x80); goto CONV_END; -conv_123x_xx2B: as_u16(dst) = ((as_u32(src) >> 8) & 0xffff) ^ 0x80; goto CONV_END; -conv_123x_xB21: as_u32(dst) = swab32(as_u32(src)) ^ 0x800000; goto CONV_END; -conv_123x_12Bx: as_u32(dst) = as_u32(src) ^ 0x8000; goto CONV_END; -conv_123x_B210: as_u32(dst) = swab32(as_u32(src) ^ 0x8000) << 8; goto CONV_END; -conv_123x_012B: as_u32(dst) = (as_u32(src) >> 8) ^ 0x80; goto CONV_END; -conv_1234_xxx1: as_u8(dst) = as_u32(src) >> 24; goto CONV_END; -conv_1234_xx12: as_u16(dst) = as_u32(src) >> 16; goto CONV_END; -conv_1234_xx21: as_u16(dst) = swab16(as_u32(src) >> 16); goto CONV_END; -conv_1234_x123: as_u32(dst) = as_u32(src) >> 8; goto CONV_END; -conv_1234_321x: as_u32(dst) = swab32(as_u32(src)) << 8; goto CONV_END; -conv_1234_1234: as_u32(dst) = as_u32(src); goto CONV_END; -conv_1234_4321: as_u32(dst) = swab32(as_u32(src)); goto CONV_END; -conv_1234_xxx9: as_u8(dst) = (as_u32(src) >> 24) ^ 0x80; goto CONV_END; -conv_1234_xx92: as_u16(dst) = (as_u32(src) >> 16) ^ 0x8000; goto CONV_END; -conv_1234_xx29: as_u16(dst) = swab16(as_u32(src) >> 16) ^ 0x80; goto CONV_END; -conv_1234_x923: as_u32(dst) = (as_u32(src) >> 8) ^ 0x800000; goto CONV_END; -conv_1234_329x: as_u32(dst) = (swab32(as_u32(src)) ^ 0x80) << 8; goto CONV_END; -conv_1234_9234: as_u32(dst) = as_u32(src) ^ 0x80000000; goto CONV_END; -conv_1234_4329: as_u32(dst) = swab32(as_u32(src)) ^ 0x80; goto CONV_END; -conv_1234_xxx4: as_u8(dst) = as_u32(src) & 0xff; goto CONV_END; -conv_1234_xx43: as_u16(dst) = swab16(as_u32(src)); goto CONV_END; -conv_1234_xx34: as_u16(dst) = as_u32(src) & 0xffff; goto CONV_END; -conv_1234_x432: as_u32(dst) = swab32(as_u32(src)) >> 8; goto CONV_END; -conv_1234_234x: as_u32(dst) = as_u32(src) << 8; goto CONV_END; -conv_1234_xxxC: as_u8(dst) = (as_u32(src) & 0xff) ^ 0x80; goto CONV_END; -conv_1234_xxC3: as_u16(dst) = swab16(as_u32(src) ^ 0x80); goto CONV_END; -conv_1234_xx3C: as_u16(dst) = (as_u32(src) & 0xffff) ^ 0x80; goto CONV_END; -conv_1234_xC32: as_u32(dst) = (swab32(as_u32(src)) >> 8) ^ 0x800000; goto CONV_END; -conv_1234_23Cx: as_u32(dst) = (as_u32(src) ^ 0x80) << 8; goto CONV_END; -conv_1234_C321: as_u32(dst) = swab32(as_u32(src) ^ 0x80); goto CONV_END; -conv_1234_123C: as_u32(dst) = as_u32(src) ^ 0x80; goto CONV_END; -} -#endif - -#ifdef GET_S16_LABELS -/* src_wid src_endswap unsigned */ -static void *get_s16_labels[4 * 2 * 2] = { - &&get_s16_xxx1_xx10, /* 8h -> 16h */ - &&get_s16_xxx1_xx90, /* 8h ^> 16h */ - &&get_s16_xxx1_xx10, /* 8s -> 16h */ - &&get_s16_xxx1_xx90, /* 8s ^> 16h */ - &&get_s16_xx12_xx12, /* 16h -> 16h */ - &&get_s16_xx12_xx92, /* 16h ^> 16h */ - &&get_s16_xx12_xx21, /* 16s -> 16h */ - &&get_s16_xx12_xxA1, /* 16s ^> 16h */ - &&get_s16_x123_xx12, /* 24h -> 16h */ - &&get_s16_x123_xx92, /* 24h ^> 16h */ - &&get_s16_123x_xx32, /* 24s -> 16h */ - &&get_s16_123x_xxB2, /* 24s ^> 16h */ - &&get_s16_1234_xx12, /* 32h -> 16h */ - &&get_s16_1234_xx92, /* 32h ^> 16h */ - &&get_s16_1234_xx43, /* 32s -> 16h */ - &&get_s16_1234_xxC3, /* 32s ^> 16h */ -}; -#endif - -#ifdef GET_S16_END -while(0) { -get_s16_xxx1_xx10: sample = (u_int16_t)as_u8(src) << 8; goto GET_S16_END; -get_s16_xxx1_xx90: sample = (u_int16_t)(as_u8(src) ^ 0x80) << 8; goto GET_S16_END; -get_s16_xx12_xx12: sample = as_u16(src); goto GET_S16_END; -get_s16_xx12_xx92: sample = as_u16(src) ^ 0x8000; goto GET_S16_END; -get_s16_xx12_xx21: sample = swab16(as_u16(src)); goto GET_S16_END; -get_s16_xx12_xxA1: sample = swab16(as_u16(src) ^ 0x80); goto GET_S16_END; -get_s16_x123_xx12: sample = as_u32(src) >> 8; goto GET_S16_END; -get_s16_x123_xx92: sample = (as_u32(src) >> 8) ^ 0x8000; goto GET_S16_END; -get_s16_123x_xx32: sample = swab16(as_u32(src) >> 8); goto GET_S16_END; -get_s16_123x_xxB2: sample = swab16((as_u32(src) >> 8) ^ 0x8000); goto GET_S16_END; -get_s16_1234_xx12: sample = as_u32(src) >> 16; goto GET_S16_END; -get_s16_1234_xx92: sample = (as_u32(src) >> 16) ^ 0x8000; goto GET_S16_END; -get_s16_1234_xx43: sample = swab16(as_u32(src)); goto GET_S16_END; -get_s16_1234_xxC3: sample = swab16(as_u32(src) ^ 0x80); goto GET_S16_END; -} -#endif - -#ifdef PUT_S16_LABELS -/* dst_wid dst_endswap unsigned */ -static void *put_s16_labels[4 * 2 * 2] = { - &&put_s16_xx12_xxx1, /* 16h -> 8h */ - &&put_s16_xx12_xxx9, /* 16h ^> 8h */ - &&put_s16_xx12_xxx1, /* 16h -> 8s */ - &&put_s16_xx12_xxx9, /* 16h ^> 8s */ - &&put_s16_xx12_xx12, /* 16h -> 16h */ - &&put_s16_xx12_xx92, /* 16h ^> 16h */ - &&put_s16_xx12_xx21, /* 16h -> 16s */ - &&put_s16_xx12_xx29, /* 16h ^> 16s */ - &&put_s16_xx12_x120, /* 16h -> 24h */ - &&put_s16_xx12_x920, /* 16h ^> 24h */ - &&put_s16_xx12_021x, /* 16h -> 24s */ - &&put_s16_xx12_029x, /* 16h ^> 24s */ - &&put_s16_xx12_1200, /* 16h -> 32h */ - &&put_s16_xx12_9200, /* 16h ^> 32h */ - &&put_s16_xx12_0021, /* 16h -> 32s */ - &&put_s16_xx12_0029, /* 16h ^> 32s */ -}; -#endif - -#ifdef PUT_S16_END -while (0) { -put_s16_xx12_xxx1: as_u8(dst) = sample >> 8; goto PUT_S16_END; -put_s16_xx12_xxx9: as_u8(dst) = (sample >> 8) ^ 0x80; goto PUT_S16_END; -put_s16_xx12_xx12: as_u16(dst) = sample; goto PUT_S16_END; -put_s16_xx12_xx92: as_u16(dst) = sample ^ 0x8000; goto PUT_S16_END; -put_s16_xx12_xx21: as_u16(dst) = swab16(sample); goto PUT_S16_END; -put_s16_xx12_xx29: as_u16(dst) = swab16(sample) ^ 0x80; goto PUT_S16_END; -put_s16_xx12_x120: as_u32(dst) = (u_int32_t)sample << 8; goto PUT_S16_END; -put_s16_xx12_x920: as_u32(dst) = (u_int32_t)(sample ^ 0x8000) << 8; goto PUT_S16_END; -put_s16_xx12_021x: as_u32(dst) = (u_int32_t)swab16(sample) << 8; goto PUT_S16_END; -put_s16_xx12_029x: as_u32(dst) = (u_int32_t)(swab16(sample) ^ 0x80) << 8; goto PUT_S16_END; -put_s16_xx12_1200: as_u32(dst) = (u_int32_t)sample << 16; goto PUT_S16_END; -put_s16_xx12_9200: as_u32(dst) = (u_int32_t)(sample ^ 0x8000) << 16; goto PUT_S16_END; -put_s16_xx12_0021: as_u32(dst) = (u_int32_t)swab16(sample); goto PUT_S16_END; -put_s16_xx12_0029: as_u32(dst) = (u_int32_t)swab16(sample) ^ 0x80; goto PUT_S16_END; -} -#endif - -#undef as_u8 -#undef as_u16 -#undef as_u32 -#undef as_s8 -#undef as_s16 -#undef as_s32 diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c index 18d8a0f4e816..9eb267913c38 100644 --- a/sound/core/oss/rate.c +++ b/sound/core/oss/rate.c @@ -1,6 +1,6 @@ /* * Rate conversion Plug-In - * Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz> * * * This library is free software; you can redistribute it and/or modify @@ -20,9 +20,6 @@ */ #include <sound/driver.h> - -#ifdef CONFIG_SND_PCM_OSS_PLUGINS - #include <linux/time.h> #include <sound/core.h> #include <sound/pcm.h> @@ -340,5 +337,3 @@ int snd_pcm_plugin_build_rate(struct snd_pcm_substream *plug, *r_plugin = plugin; return 0; } - -#endif diff --git a/sound/core/oss/route.c b/sound/core/oss/route.c index 46917dc0196b..de3ffdeaf7e3 100644 --- a/sound/core/oss/route.c +++ b/sound/core/oss/route.c @@ -20,9 +20,6 @@ */ #include <sound/driver.h> - -#ifdef CONFIG_SND_PCM_OSS_PLUGINS - #include <linux/slab.h> #include <linux/time.h> #include <sound/core.h> @@ -108,5 +105,3 @@ int snd_pcm_plugin_build_route(struct snd_pcm_substream *plug, *r_plugin = plugin; return 0; } - -#endif diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 2743414fc8fa..cf9b9493d41d 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -1,6 +1,6 @@ /* * Digital Audio (PCM) abstract layer - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -30,7 +30,7 @@ #include <sound/control.h> #include <sound/info.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>, Abramo Bagnara <abramo@alsa-project.org>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, Abramo Bagnara <abramo@alsa-project.org>"); MODULE_DESCRIPTION("Midlevel PCM code for ALSA."); MODULE_LICENSE("GPL"); diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 9fefcaa2c324..806f1fba5446 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1,6 +1,6 @@ /* * Digital Audio (PCM) abstract layer - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Abramo Bagnara <abramo@alsa-project.org> * * diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 95b1b2f0b1e2..a13e38cfd2c6 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -1,6 +1,6 @@ /* * Digital Audio (PCM) abstract layer - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index 0019c59a779d..dd9aa51d8c82 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -1,6 +1,6 @@ /* * PCM Interface - misc routines - * Copyright (c) 1998 by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) 1998 by Jaroslav Kysela <perex@perex.cz> * * * This library is free software; you can redistribute it and/or modify @@ -422,38 +422,6 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int EXPORT_SYMBOL(snd_pcm_format_set_silence); -/* [width][unsigned][bigendian] */ -static int linear_formats[4][2][2] = { - {{ SNDRV_PCM_FORMAT_S8, SNDRV_PCM_FORMAT_S8}, - { SNDRV_PCM_FORMAT_U8, SNDRV_PCM_FORMAT_U8}}, - {{SNDRV_PCM_FORMAT_S16_LE, SNDRV_PCM_FORMAT_S16_BE}, - {SNDRV_PCM_FORMAT_U16_LE, SNDRV_PCM_FORMAT_U16_BE}}, - {{SNDRV_PCM_FORMAT_S24_LE, SNDRV_PCM_FORMAT_S24_BE}, - {SNDRV_PCM_FORMAT_U24_LE, SNDRV_PCM_FORMAT_U24_BE}}, - {{SNDRV_PCM_FORMAT_S32_LE, SNDRV_PCM_FORMAT_S32_BE}, - {SNDRV_PCM_FORMAT_U32_LE, SNDRV_PCM_FORMAT_U32_BE}} -}; - -/** - * snd_pcm_build_linear_format - return the suitable linear format for the given condition - * @width: the bit-width - * @unsignd: 1 if unsigned, 0 if signed. - * @big_endian: 1 if big-endian, 0 if little-endian - * - * Returns the suitable linear format for the given condition. - */ -snd_pcm_format_t snd_pcm_build_linear_format(int width, int unsignd, int big_endian) -{ - if (width & 7) - return SND_PCM_FORMAT_UNKNOWN; - width = (width / 8) - 1; - if (width < 0 || width >= 4) - return SND_PCM_FORMAT_UNKNOWN; - return linear_formats[width][!!unsignd][!!big_endian]; -} - -EXPORT_SYMBOL(snd_pcm_build_linear_format); - /** * snd_pcm_limit_hw_rates - determine rate_min/rate_max fields * @runtime: the runtime instance @@ -465,21 +433,16 @@ EXPORT_SYMBOL(snd_pcm_build_linear_format); */ int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime) { - static unsigned rates[] = { - /* ATTENTION: these values depend on the definition in pcm.h! */ - 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, - 64000, 88200, 96000, 176400, 192000 - }; int i; - for (i = 0; i < (int)ARRAY_SIZE(rates); i++) { + for (i = 0; i < (int)snd_pcm_known_rates.count; i++) { if (runtime->hw.rates & (1 << i)) { - runtime->hw.rate_min = rates[i]; + runtime->hw.rate_min = snd_pcm_known_rates.list[i]; break; } } - for (i = (int)ARRAY_SIZE(rates) - 1; i >= 0; i--) { + for (i = (int)snd_pcm_known_rates.count - 1; i >= 0; i--) { if (runtime->hw.rates & (1 << i)) { - runtime->hw.rate_max = rates[i]; + runtime->hw.rate_max = snd_pcm_known_rates.list[i]; break; } } @@ -487,3 +450,21 @@ int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime) } EXPORT_SYMBOL(snd_pcm_limit_hw_rates); + +/** + * snd_pcm_rate_to_rate_bit - converts sample rate to SNDRV_PCM_RATE_xxx bit + * @rate: the sample rate to convert + * + * Returns the SNDRV_PCM_RATE_xxx flag that corresponds to the given rate, or + * SNDRV_PCM_RATE_KNOT for an unknown rate. + */ +unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate) +{ + unsigned int i; + + for (i = 0; i < snd_pcm_known_rates.count; i++) + if (snd_pcm_known_rates.list[i] == rate) + return 1u << i; + return SNDRV_PCM_RATE_KNOT; +} +EXPORT_SYMBOL(snd_pcm_rate_to_rate_bit); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 59b29cd482ae..fb3dde4db045 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1,6 +1,6 @@ /* * Digital Audio (PCM) abstract layer - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -1787,12 +1787,18 @@ static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params, static unsigned int rates[] = { 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000 }; +const struct snd_pcm_hw_constraint_list snd_pcm_known_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, +}; + static int snd_pcm_hw_rule_rate(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { struct snd_pcm_hardware *hw = rule->private; return snd_interval_list(hw_param_interval(params, rule->var), - ARRAY_SIZE(rates), rates, hw->rates); + snd_pcm_known_rates.count, + snd_pcm_known_rates.list, hw->rates); } static int snd_pcm_hw_rule_buffer_bytes_max(struct snd_pcm_hw_params *params, diff --git a/sound/core/pcm_timer.c b/sound/core/pcm_timer.c index d94ed16d21ea..23aa9a27e215 100644 --- a/sound/core/pcm_timer.c +++ b/sound/core/pcm_timer.c @@ -1,6 +1,6 @@ /* * Digital Audio (PCM) abstract layer - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index e470c3c7d611..b8e700b94e59 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -1,6 +1,6 @@ /* * Abstract layer for MIDI v1.0 stream - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -30,14 +30,13 @@ #include <linux/mutex.h> #include <linux/moduleparam.h> #include <linux/delay.h> -#include <linux/wait.h> #include <sound/rawmidi.h> #include <sound/info.h> #include <sound/control.h> #include <sound/minors.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Midlevel RawMidi code for ALSA."); MODULE_LICENSE("GPL"); diff --git a/sound/core/seq/Makefile b/sound/core/seq/Makefile index 402e2b4a34c6..ceef14afee30 100644 --- a/sound/core/seq/Makefile +++ b/sound/core/seq/Makefile @@ -1,6 +1,6 @@ # # Makefile for ALSA -# Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz> +# Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz> # obj-$(CONFIG_SND) += instr/ diff --git a/sound/core/seq/instr/Makefile b/sound/core/seq/instr/Makefile index 69138f30a293..608960364813 100644 --- a/sound/core/seq/instr/Makefile +++ b/sound/core/seq/instr/Makefile @@ -1,6 +1,6 @@ # # Makefile for ALSA -# Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz> +# Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz> # snd-ainstr-fm-objs := ainstr_fm.o diff --git a/sound/core/seq/instr/ainstr_gf1.c b/sound/core/seq/instr/ainstr_gf1.c index c640e1cf854d..49400262b1eb 100644 --- a/sound/core/seq/instr/ainstr_gf1.c +++ b/sound/core/seq/instr/ainstr_gf1.c @@ -1,6 +1,6 @@ /* * GF1 (GUS) Patch - Instrument routines - * Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz> * * 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 @@ -26,7 +26,7 @@ #include <sound/initval.h> #include <asm/uaccess.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Advanced Linux Sound Architecture GF1 (GUS) Patch support."); MODULE_LICENSE("GPL"); diff --git a/sound/core/seq/instr/ainstr_iw.c b/sound/core/seq/instr/ainstr_iw.c index 5367baee2d08..6c40eb73fa9f 100644 --- a/sound/core/seq/instr/ainstr_iw.c +++ b/sound/core/seq/instr/ainstr_iw.c @@ -1,6 +1,6 @@ /* * IWFFFF - AMD InterWave (tm) - Instrument routines - * Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz> * * 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 @@ -26,7 +26,7 @@ #include <sound/initval.h> #include <asm/uaccess.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Advanced Linux Sound Architecture IWFFFF support."); MODULE_LICENSE("GPL"); diff --git a/sound/core/seq/instr/ainstr_simple.c b/sound/core/seq/instr/ainstr_simple.c index ac717bef9d77..78f68bee24fe 100644 --- a/sound/core/seq/instr/ainstr_simple.c +++ b/sound/core/seq/instr/ainstr_simple.c @@ -1,6 +1,6 @@ /* * Simple (MOD player) - Instrument routines - * Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz> * * 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 @@ -26,7 +26,7 @@ #include <sound/initval.h> #include <asm/uaccess.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Advanced Linux Sound Architecture Simple Instrument support."); MODULE_LICENSE("GPL"); diff --git a/sound/core/seq/oss/Makefile b/sound/core/seq/oss/Makefile index a37ddedf7107..b38406b8463c 100644 --- a/sound/core/seq/oss/Makefile +++ b/sound/core/seq/oss/Makefile @@ -1,6 +1,6 @@ # # Makefile for ALSA -# Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz> +# Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz> # snd-seq-oss-objs := seq_oss.o seq_oss_init.o seq_oss_timer.o seq_oss_ioctl.o \ diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c index ca5a2ed4d7c3..d0d721c22eac 100644 --- a/sound/core/seq/oss/seq_oss_init.c +++ b/sound/core/seq/oss/seq_oss_init.c @@ -176,29 +176,29 @@ snd_seq_oss_open(struct file *file, int level) int i, rc; struct seq_oss_devinfo *dp; - if ((dp = kzalloc(sizeof(*dp), GFP_KERNEL)) == NULL) { + dp = kzalloc(sizeof(*dp), GFP_KERNEL); + if (!dp) { snd_printk(KERN_ERR "can't malloc device info\n"); return -ENOMEM; } debug_printk(("oss_open: dp = %p\n", dp)); + dp->cseq = system_client; + dp->port = -1; + dp->queue = -1; + for (i = 0; i < SNDRV_SEQ_OSS_MAX_CLIENTS; i++) { if (client_table[i] == NULL) break; } + + dp->index = i; if (i >= SNDRV_SEQ_OSS_MAX_CLIENTS) { snd_printk(KERN_ERR "too many applications\n"); - kfree(dp); - return -ENOMEM; + rc = -ENOMEM; + goto _error; } - dp->index = i; - dp->cseq = system_client; - dp->port = -1; - dp->queue = -1; - dp->readq = NULL; - dp->writeq = NULL; - /* look up synth and midi devices */ snd_seq_oss_synth_setup(dp); snd_seq_oss_midi_setup(dp); @@ -211,14 +211,16 @@ snd_seq_oss_open(struct file *file, int level) /* create port */ debug_printk(("create new port\n")); - if ((rc = create_port(dp)) < 0) { + rc = create_port(dp); + if (rc < 0) { snd_printk(KERN_ERR "can't create port\n"); goto _error; } /* allocate queue */ debug_printk(("allocate queue\n")); - if ((rc = alloc_seq_queue(dp)) < 0) + rc = alloc_seq_queue(dp); + if (rc < 0) goto _error; /* set address */ @@ -235,7 +237,8 @@ snd_seq_oss_open(struct file *file, int level) /* initialize read queue */ debug_printk(("initialize read queue\n")); if (is_read_mode(dp->file_mode)) { - if ((dp->readq = snd_seq_oss_readq_new(dp, maxqlen)) == NULL) { + dp->readq = snd_seq_oss_readq_new(dp, maxqlen); + if (!dp->readq) { rc = -ENOMEM; goto _error; } @@ -245,7 +248,7 @@ snd_seq_oss_open(struct file *file, int level) debug_printk(("initialize write queue\n")); if (is_write_mode(dp->file_mode)) { dp->writeq = snd_seq_oss_writeq_new(dp, maxqlen); - if (dp->writeq == NULL) { + if (!dp->writeq) { rc = -ENOMEM; goto _error; } @@ -253,7 +256,8 @@ snd_seq_oss_open(struct file *file, int level) /* initialize timer */ debug_printk(("initialize timer\n")); - if ((dp->timer = snd_seq_oss_timer_new(dp)) == NULL) { + dp->timer = snd_seq_oss_timer_new(dp); + if (!dp->timer) { snd_printk(KERN_ERR "can't alloc timer\n"); rc = -ENOMEM; goto _error; @@ -276,11 +280,13 @@ snd_seq_oss_open(struct file *file, int level) return 0; _error: + snd_seq_oss_writeq_delete(dp->writeq); + snd_seq_oss_readq_delete(dp->readq); snd_seq_oss_synth_cleanup(dp); snd_seq_oss_midi_cleanup(dp); - i = dp->queue; delete_port(dp); - delete_seq_queue(i); + delete_seq_queue(dp->queue); + kfree(dp); return rc; } diff --git a/sound/core/seq/oss/seq_oss_writeq.c b/sound/core/seq/oss/seq_oss_writeq.c index 5c8495601a38..217424858191 100644 --- a/sound/core/seq/oss/seq_oss_writeq.c +++ b/sound/core/seq/oss/seq_oss_writeq.c @@ -63,8 +63,10 @@ snd_seq_oss_writeq_new(struct seq_oss_devinfo *dp, int maxlen) void snd_seq_oss_writeq_delete(struct seq_oss_writeq *q) { - snd_seq_oss_writeq_clear(q); /* to be sure */ - kfree(q); + if (q) { + snd_seq_oss_writeq_clear(q); /* to be sure */ + kfree(q); + } } diff --git a/sound/core/seq/seq.c b/sound/core/seq/seq.c index 2f0d8773ac6b..1878208a8026 100644 --- a/sound/core/seq/seq.c +++ b/sound/core/seq/seq.c @@ -53,7 +53,7 @@ int seq_default_timer_device = int seq_default_timer_subdevice = 0; int seq_default_timer_resolution = 0; /* Hz */ -MODULE_AUTHOR("Frank van de Pol <fvdpol@coil.demon.nl>, Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Frank van de Pol <fvdpol@coil.demon.nl>, Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Advanced Linux Sound Architecture sequencer."); MODULE_LICENSE("GPL"); diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index b31b5282a2c8..2e3fa25ab19f 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -1,7 +1,7 @@ /* * ALSA sequencer Client Manager * Copyright (c) 1998-2001 by Frank van de Pol <fvdpol@coil.demon.nl> - * Jaroslav Kysela <perex@suse.cz> + * Jaroslav Kysela <perex@perex.cz> * Takashi Iwai <tiwai@suse.de> * * diff --git a/sound/core/seq/seq_instr.c b/sound/core/seq/seq_instr.c index 5efe6523a589..9a6fd56c9109 100644 --- a/sound/core/seq/seq_instr.c +++ b/sound/core/seq/seq_instr.c @@ -1,6 +1,6 @@ /* * Generic Instrument routines for ALSA sequencer - * Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz> * * 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 @@ -26,7 +26,7 @@ #include <sound/seq_instr.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Advanced Linux Sound Architecture sequencer instrument library."); MODULE_LICENSE("GPL"); @@ -109,7 +109,7 @@ void snd_seq_instr_list_free(struct snd_seq_kinstr_list **list_ptr) spin_lock_irqsave(&list->lock, flags); while (instr->use) { spin_unlock_irqrestore(&list->lock, flags); - schedule_timeout(1); + schedule_timeout_uninterruptible(1); spin_lock_irqsave(&list->lock, flags); } spin_unlock_irqrestore(&list->lock, flags); @@ -198,8 +198,10 @@ int snd_seq_instr_list_free_cond(struct snd_seq_kinstr_list *list, while (flist) { instr = flist; flist = instr->next; - while (instr->use) - schedule_timeout(1); + while (instr->use) { + schedule_timeout_uninterruptible(1); + barrier(); + } if (snd_seq_instr_free(instr, atomic)<0) snd_printk(KERN_WARNING "instrument free problem\n"); instr = next; @@ -555,7 +557,7 @@ static int instr_free(struct snd_seq_kinstr_ops *ops, SNDRV_SEQ_INSTR_NOTIFY_REMOVE); while (instr->use) { spin_unlock_irqrestore(&list->lock, flags); - schedule_timeout(1); + schedule_timeout_uninterruptible(1); spin_lock_irqsave(&list->lock, flags); } spin_unlock_irqrestore(&list->lock, flags); diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c index a3dc5e01e9f2..a72a1945bf8a 100644 --- a/sound/core/seq/seq_memory.c +++ b/sound/core/seq/seq_memory.c @@ -1,7 +1,7 @@ /* * ALSA sequencer Memory Manager * Copyright (c) 1998 by Frank van de Pol <fvdpol@coil.demon.nl> - * Jaroslav Kysela <perex@suse.cz> + * Jaroslav Kysela <perex@perex.cz> * 2000 by Takashi Iwai <tiwai@suse.de> * * This program is free software; you can redistribute it and/or modify diff --git a/sound/core/seq/seq_midi.c b/sound/core/seq/seq_midi.c index 1daa5b069c79..5929aaf1df9d 100644 --- a/sound/core/seq/seq_midi.c +++ b/sound/core/seq/seq_midi.c @@ -1,7 +1,7 @@ /* * Generic MIDI synth driver for ALSA sequencer * Copyright (c) 1998 by Frank van de Pol <fvdpol@coil.demon.nl> - * Jaroslav Kysela <perex@suse.cz> + * Jaroslav Kysela <perex@perex.cz> * * 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 @@ -40,7 +40,7 @@ Possible options for midisynth module: #include <sound/seq_midi_event.h> #include <sound/initval.h> -MODULE_AUTHOR("Frank van de Pol <fvdpol@coil.demon.nl>, Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Frank van de Pol <fvdpol@coil.demon.nl>, Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Advanced Linux Sound Architecture sequencer MIDI synth."); MODULE_LICENSE("GPL"); static int output_buffer_size = PAGE_SIZE; diff --git a/sound/core/seq/seq_midi_event.c b/sound/core/seq/seq_midi_event.c index 5ff80b776906..b6820a5a73fc 100644 --- a/sound/core/seq/seq_midi_event.c +++ b/sound/core/seq/seq_midi_event.c @@ -2,7 +2,7 @@ * MIDI byte <-> sequencer event coder * * Copyright (C) 1998,99 Takashi Iwai <tiwai@suse.de>, - * Jaroslav Kysela <perex@suse.cz> + * Jaroslav Kysela <perex@perex.cz> * * 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 @@ -28,14 +28,13 @@ #include <sound/seq_midi_event.h> #include <sound/asoundef.h> -MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>, Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>, Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("MIDI byte <-> sequencer event coder"); MODULE_LICENSE("GPL"); -/* queue type */ -/* from 0 to 7 are normal commands (note off, on, etc.) */ -#define ST_NOTEOFF 0 -#define ST_NOTEON 1 +/* event type, index into status_event[] */ +/* from 0 to 6 are normal commands (note off, on, etc.) for 0x9?-0xe? */ +#define ST_INVALID 7 #define ST_SPECIAL 8 #define ST_SYSEX ST_SPECIAL /* from 8 to 15 are events for 0xf0-0xf7 */ @@ -65,32 +64,33 @@ static struct status_event_list { void (*encode)(struct snd_midi_event *dev, struct snd_seq_event *ev); void (*decode)(struct snd_seq_event *ev, unsigned char *buf); } status_event[] = { - /* 0x80 - 0xf0 */ - {SNDRV_SEQ_EVENT_NOTEOFF, 2, note_event, note_decode}, - {SNDRV_SEQ_EVENT_NOTEON, 2, note_event, note_decode}, - {SNDRV_SEQ_EVENT_KEYPRESS, 2, note_event, note_decode}, - {SNDRV_SEQ_EVENT_CONTROLLER, 2, two_param_ctrl_event, two_param_decode}, - {SNDRV_SEQ_EVENT_PGMCHANGE, 1, one_param_ctrl_event, one_param_decode}, - {SNDRV_SEQ_EVENT_CHANPRESS, 1, one_param_ctrl_event, one_param_decode}, - {SNDRV_SEQ_EVENT_PITCHBEND, 2, pitchbend_ctrl_event, pitchbend_decode}, - {SNDRV_SEQ_EVENT_NONE, 0, NULL, NULL}, /* 0xf0 */ + /* 0x80 - 0xef */ + {SNDRV_SEQ_EVENT_NOTEOFF, 2, note_event, note_decode}, + {SNDRV_SEQ_EVENT_NOTEON, 2, note_event, note_decode}, + {SNDRV_SEQ_EVENT_KEYPRESS, 2, note_event, note_decode}, + {SNDRV_SEQ_EVENT_CONTROLLER, 2, two_param_ctrl_event, two_param_decode}, + {SNDRV_SEQ_EVENT_PGMCHANGE, 1, one_param_ctrl_event, one_param_decode}, + {SNDRV_SEQ_EVENT_CHANPRESS, 1, one_param_ctrl_event, one_param_decode}, + {SNDRV_SEQ_EVENT_PITCHBEND, 2, pitchbend_ctrl_event, pitchbend_decode}, + /* invalid */ + {SNDRV_SEQ_EVENT_NONE, -1, NULL, NULL}, /* 0xf0 - 0xff */ - {SNDRV_SEQ_EVENT_SYSEX, 1, NULL, NULL}, /* sysex: 0xf0 */ - {SNDRV_SEQ_EVENT_QFRAME, 1, one_param_event, one_param_decode}, /* 0xf1 */ - {SNDRV_SEQ_EVENT_SONGPOS, 2, songpos_event, songpos_decode}, /* 0xf2 */ - {SNDRV_SEQ_EVENT_SONGSEL, 1, one_param_event, one_param_decode}, /* 0xf3 */ - {SNDRV_SEQ_EVENT_NONE, 0, NULL, NULL}, /* 0xf4 */ - {SNDRV_SEQ_EVENT_NONE, 0, NULL, NULL}, /* 0xf5 */ - {SNDRV_SEQ_EVENT_TUNE_REQUEST, 0, NULL, NULL}, /* 0xf6 */ - {SNDRV_SEQ_EVENT_NONE, 0, NULL, NULL}, /* 0xf7 */ - {SNDRV_SEQ_EVENT_CLOCK, 0, NULL, NULL}, /* 0xf8 */ - {SNDRV_SEQ_EVENT_NONE, 0, NULL, NULL}, /* 0xf9 */ - {SNDRV_SEQ_EVENT_START, 0, NULL, NULL}, /* 0xfa */ - {SNDRV_SEQ_EVENT_CONTINUE, 0, NULL, NULL}, /* 0xfb */ - {SNDRV_SEQ_EVENT_STOP, 0, NULL, NULL}, /* 0xfc */ - {SNDRV_SEQ_EVENT_NONE, 0, NULL, NULL}, /* 0xfd */ - {SNDRV_SEQ_EVENT_SENSING, 0, NULL, NULL}, /* 0xfe */ - {SNDRV_SEQ_EVENT_RESET, 0, NULL, NULL}, /* 0xff */ + {SNDRV_SEQ_EVENT_SYSEX, 1, NULL, NULL}, /* sysex: 0xf0 */ + {SNDRV_SEQ_EVENT_QFRAME, 1, one_param_event, one_param_decode}, /* 0xf1 */ + {SNDRV_SEQ_EVENT_SONGPOS, 2, songpos_event, songpos_decode}, /* 0xf2 */ + {SNDRV_SEQ_EVENT_SONGSEL, 1, one_param_event, one_param_decode}, /* 0xf3 */ + {SNDRV_SEQ_EVENT_NONE, -1, NULL, NULL}, /* 0xf4 */ + {SNDRV_SEQ_EVENT_NONE, -1, NULL, NULL}, /* 0xf5 */ + {SNDRV_SEQ_EVENT_TUNE_REQUEST, 0, NULL, NULL}, /* 0xf6 */ + {SNDRV_SEQ_EVENT_NONE, -1, NULL, NULL}, /* 0xf7 */ + {SNDRV_SEQ_EVENT_CLOCK, 0, NULL, NULL}, /* 0xf8 */ + {SNDRV_SEQ_EVENT_NONE, -1, NULL, NULL}, /* 0xf9 */ + {SNDRV_SEQ_EVENT_START, 0, NULL, NULL}, /* 0xfa */ + {SNDRV_SEQ_EVENT_CONTINUE, 0, NULL, NULL}, /* 0xfb */ + {SNDRV_SEQ_EVENT_STOP, 0, NULL, NULL}, /* 0xfc */ + {SNDRV_SEQ_EVENT_NONE, -1, NULL, NULL}, /* 0xfd */ + {SNDRV_SEQ_EVENT_SENSING, 0, NULL, NULL}, /* 0xfe */ + {SNDRV_SEQ_EVENT_RESET, 0, NULL, NULL}, /* 0xff */ }; static int extra_decode_ctrl14(struct snd_midi_event *dev, unsigned char *buf, int len, @@ -129,6 +129,7 @@ int snd_midi_event_new(int bufsize, struct snd_midi_event **rdev) } dev->bufsize = bufsize; dev->lastcmd = 0xff; + dev->type = ST_INVALID; spin_lock_init(&dev->lock); *rdev = dev; return 0; @@ -149,7 +150,7 @@ static inline void reset_encode(struct snd_midi_event *dev) { dev->read = 0; dev->qlen = 0; - dev->type = 0; + dev->type = ST_INVALID; } void snd_midi_event_reset_encode(struct snd_midi_event *dev) @@ -251,29 +252,31 @@ int snd_midi_event_encode_byte(struct snd_midi_event *dev, int c, ev->type = status_event[ST_SPECIAL + c - 0xf0].event; ev->flags &= ~SNDRV_SEQ_EVENT_LENGTH_MASK; ev->flags |= SNDRV_SEQ_EVENT_LENGTH_FIXED; - return 1; + return ev->type != SNDRV_SEQ_EVENT_NONE; } spin_lock_irqsave(&dev->lock, flags); - if (dev->qlen > 0) { - /* rest of command */ - dev->buf[dev->read++] = c; - if (dev->type != ST_SYSEX) - dev->qlen--; - } else { + if ((c & 0x80) && + (c != MIDI_CMD_COMMON_SYSEX_END || dev->type != ST_SYSEX)) { /* new command */ + dev->buf[0] = c; + if ((c & 0xf0) == 0xf0) /* system messages */ + dev->type = (c & 0x0f) + ST_SPECIAL; + else + dev->type = (c >> 4) & 0x07; dev->read = 1; - if (c & 0x80) { - dev->buf[0] = c; - if ((c & 0xf0) == 0xf0) /* special events */ - dev->type = (c & 0x0f) + ST_SPECIAL; - else - dev->type = (c >> 4) & 0x07; - dev->qlen = status_event[dev->type].qlen; - } else { - /* process this byte as argument */ + dev->qlen = status_event[dev->type].qlen; + } else { + if (dev->qlen > 0) { + /* rest of command */ dev->buf[dev->read++] = c; + if (dev->type != ST_SYSEX) + dev->qlen--; + } else { + /* running status */ + dev->buf[1] = c; dev->qlen = status_event[dev->type].qlen - 1; + dev->read = 2; } } if (dev->qlen == 0) { @@ -282,6 +285,8 @@ int snd_midi_event_encode_byte(struct snd_midi_event *dev, int c, ev->flags |= SNDRV_SEQ_EVENT_LENGTH_FIXED; if (status_event[dev->type].encode) /* set data values */ status_event[dev->type].encode(dev, ev); + if (dev->type >= ST_SPECIAL) + dev->type = ST_INVALID; rc = 1; } else if (dev->type == ST_SYSEX) { if (c == MIDI_CMD_COMMON_SYSEX_END || diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index eefd1cf872b4..b6e23ad12ab9 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -1,7 +1,7 @@ /* * ALSA sequencer Ports * Copyright (c) 1998 by Frank van de Pol <fvdpol@coil.demon.nl> - * Jaroslav Kysela <perex@suse.cz> + * Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c index b4b9a132cb16..8716352afc81 100644 --- a/sound/core/seq/seq_timer.c +++ b/sound/core/seq/seq_timer.c @@ -1,7 +1,7 @@ /* * ALSA sequencer Timer * Copyright (c) 1998-1999 by Frank van de Pol <fvdpol@coil.demon.nl> - * Jaroslav Kysela <perex@suse.cz> + * Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/sound/core/sound.c b/sound/core/sound.c index 8dc7a3b32b98..7b486c4d70db 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -1,6 +1,6 @@ /* * Advanced Linux Sound Architecture - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -42,7 +42,7 @@ EXPORT_SYMBOL(snd_major); static int cards_limit = 1; -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Advanced Linux Sound Architecture driver for soundcards."); MODULE_LICENSE("GPL"); module_param(major, int, 0444); @@ -266,6 +266,14 @@ int snd_register_device_for_dev(int type, struct snd_card *card, int dev, snd_minors[minor] = preg; preg->dev = device_create(sound_class, device, MKDEV(major, minor), "%s", name); + if (IS_ERR(preg->dev)) { + snd_minors[minor] = NULL; + mutex_unlock(&sound_mutex); + minor = PTR_ERR(preg->dev); + kfree(preg); + return minor; + } + if (preg->dev) dev_set_drvdata(preg->dev, private_data); diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c index 4566df41912a..dc73313b733a 100644 --- a/sound/core/sound_oss.c +++ b/sound/core/sound_oss.c @@ -1,6 +1,6 @@ /* * Advanced Linux Sound Architecture - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/sound/core/timer.c b/sound/core/timer.c index f2bbacedd567..e7dc56ca4b97 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1,6 +1,6 @@ /* * Timers abstract layer - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -44,7 +44,7 @@ #endif static int timer_limit = DEFAULT_TIMER_LIMIT; -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>, Takashi Iwai <tiwai@suse.de>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, Takashi Iwai <tiwai@suse.de>"); MODULE_DESCRIPTION("ALSA timer interface"); MODULE_LICENSE("GPL"); module_param(timer_limit, int, 0444); diff --git a/sound/drivers/Makefile b/sound/drivers/Makefile index 04112642611a..80aeff5ccdea 100644 --- a/sound/drivers/Makefile +++ b/sound/drivers/Makefile @@ -1,6 +1,6 @@ # # Makefile for ALSA -# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz> +# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz> # snd-dummy-objs := dummy.o diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 4360ae9de19c..e008f3c58eac 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -1,6 +1,6 @@ /* * Dummy soundcard - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * 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 @@ -34,7 +34,7 @@ #include <sound/rawmidi.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Dummy soundcard (/dev/null)"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{ALSA,Dummy soundcard}}"); @@ -510,15 +510,7 @@ static const DECLARE_TLV_DB_SCALE(db_scale_dummy, -4500, 30, 0); .get = snd_dummy_capsrc_get, .put = snd_dummy_capsrc_put, \ .private_value = addr } -static int snd_dummy_capsrc_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_dummy_capsrc_info snd_ctl_boolean_stereo_info static int snd_dummy_capsrc_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff --git a/sound/drivers/mpu401/Makefile b/sound/drivers/mpu401/Makefile index 3fe185d19ae5..918f83f34c11 100644 --- a/sound/drivers/mpu401/Makefile +++ b/sound/drivers/mpu401/Makefile @@ -1,6 +1,6 @@ # # Makefile for ALSA -# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz> +# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz> # snd-mpu401-objs := mpu401.o diff --git a/sound/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c index 67c6e9745418..1fc95dadde1d 100644 --- a/sound/drivers/mpu401/mpu401.c +++ b/sound/drivers/mpu401/mpu401.c @@ -1,6 +1,6 @@ /* * Driver for generic MPU-401 boards (UART mode only) - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Copyright (c) 2004 by Castet Matthieu <castet.matthieu@free.fr> * * @@ -30,7 +30,7 @@ #include <sound/mpu401.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("MPU-401 UART"); MODULE_LICENSE("GPL"); @@ -70,6 +70,9 @@ static int snd_mpu401_create(int dev, struct snd_card **rcard) struct snd_card *card; int err; + if (!uart_enter[dev]) + snd_printk(KERN_ERR "the uart_enter option is obsolete; remove it\n"); + *rcard = NULL; card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); if (card == NULL) @@ -83,8 +86,7 @@ static int snd_mpu401_create(int dev, struct snd_card **rcard) strcat(card->longname, "polled"); } - err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, port[dev], - uart_enter[dev] ? 0 : MPU401_INFO_UART_ONLY, + err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, port[dev], 0, irq[dev], irq[dev] >= 0 ? IRQF_DISABLED : 0, NULL); if (err < 0) { diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c index 85aedc348e2d..3306ecd49243 100644 --- a/sound/drivers/mpu401/mpu401_uart.c +++ b/sound/drivers/mpu401/mpu401_uart.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Routines for control of MPU-401 in UART mode * * MPU-401 supports UART mode which is not capable generate transmit @@ -39,7 +39,7 @@ #include <sound/core.h> #include <sound/mpu401.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Routines for control of MPU-401 in UART mode"); MODULE_LICENSE("GPL"); @@ -270,8 +270,7 @@ static int snd_mpu401_do_reset(struct snd_mpu401 *mpu) { if (snd_mpu401_uart_cmd(mpu, MPU401_RESET, 1)) return -EIO; - if (!(mpu->info_flags & MPU401_INFO_UART_ONLY) && - snd_mpu401_uart_cmd(mpu, MPU401_ENTER_UART, 1)) + if (snd_mpu401_uart_cmd(mpu, MPU401_ENTER_UART, 0)) return -EIO; return 0; } diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c index 2025db5947ae..911c159bb3d3 100644 --- a/sound/drivers/mts64.c +++ b/sound/drivers/mts64.c @@ -440,15 +440,7 @@ static void mts64_write_midi(struct mts64 *mts, u8 c, *********************************************************************/ /* SMPTE Switch */ -static int snd_mts64_ctl_smpte_switch_info(struct snd_kcontrol *kctl, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_mts64_ctl_smpte_switch_info snd_ctl_boolean_mono_info static int snd_mts64_ctl_smpte_switch_get(struct snd_kcontrol* kctl, struct snd_ctl_elem_value *uctl) diff --git a/sound/drivers/opl3/Makefile b/sound/drivers/opl3/Makefile index 12059785b5cb..19767a6a5c54 100644 --- a/sound/drivers/opl3/Makefile +++ b/sound/drivers/opl3/Makefile @@ -1,13 +1,11 @@ # # Makefile for ALSA -# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz> +# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz> # snd-opl3-lib-objs := opl3_lib.o opl3_synth.o -snd-opl3-synth-objs := opl3_seq.o opl3_midi.o opl3_drums.o -ifeq ($(CONFIG_SND_SEQUENCER_OSS),y) -snd-opl3-synth-objs += opl3_oss.o -endif +snd-opl3-synth-y := opl3_seq.o opl3_midi.o opl3_drums.o +snd-opl3-synth-$(CONFIG_SND_SEQUENCER_OSS) += opl3_oss.o # # this function returns: diff --git a/sound/drivers/opl3/opl3_lib.c b/sound/drivers/opl3/opl3_lib.c index 87fe376f38f0..a2b9ce060295 100644 --- a/sound/drivers/opl3/opl3_lib.c +++ b/sound/drivers/opl3/opl3_lib.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz>, + * Copyright (c) by Jaroslav Kysela <perex@perex.cz>, * Hannu Savolainen 1993-1996, * Rob Hooft * @@ -31,7 +31,7 @@ #include <linux/ioport.h> #include <sound/minors.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>, Hannu Savolainen 1993-1996, Rob Hooft"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, Hannu Savolainen 1993-1996, Rob Hooft"); MODULE_DESCRIPTION("Routines for control of AdLib FM cards (OPL2/OPL3/OPL4 chips)"); MODULE_LICENSE("GPL"); diff --git a/sound/drivers/opl4/Makefile b/sound/drivers/opl4/Makefile index 141aacbaf315..d178b39ffa60 100644 --- a/sound/drivers/opl4/Makefile +++ b/sound/drivers/opl4/Makefile @@ -1,6 +1,6 @@ # # Makefile for ALSA -# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz> +# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz> # snd-opl4-lib-objs := opl4_lib.o opl4_mixer.o opl4_proc.o diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c index d3e6a20edd38..65de3a755ddb 100644 --- a/sound/drivers/serial-u16550.c +++ b/sound/drivers/serial-u16550.c @@ -1,6 +1,6 @@ /* * serial.c - * Copyright (c) by Jaroslav Kysela <perex@suse.cz>, + * Copyright (c) by Jaroslav Kysela <perex@perex.cz>, * Isaku Yamahata <yamahata@private.email.ne.jp>, * George Hansper <ghansper@apana.org.au>, * Hannu Savolainen diff --git a/sound/drivers/vx/Makefile b/sound/drivers/vx/Makefile index 269bd8544a5d..9a168a3c1560 100644 --- a/sound/drivers/vx/Makefile +++ b/sound/drivers/vx/Makefile @@ -1,6 +1,6 @@ # # Makefile for ALSA -# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz> +# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz> # snd-vx-lib-objs := vx_core.o vx_hwdep.o vx_pcm.o vx_mixer.o vx_cmd.o vx_uer.o diff --git a/sound/drivers/vx/vx_mixer.c b/sound/drivers/vx/vx_mixer.c index f63152a6a223..b8fcd79a7e11 100644 --- a/sound/drivers/vx/vx_mixer.c +++ b/sound/drivers/vx/vx_mixer.c @@ -647,14 +647,7 @@ static int vx_audio_monitor_put(struct snd_kcontrol *kcontrol, struct snd_ctl_el return 0; } -static int vx_audio_sw_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define vx_audio_sw_info snd_ctl_boolean_stereo_info static int vx_audio_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -865,14 +858,7 @@ static int vx_peak_meter_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ return 0; } -static int vx_saturation_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define vx_saturation_info snd_ctl_boolean_stereo_info static int vx_saturation_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { diff --git a/sound/i2c/Makefile b/sound/i2c/Makefile index 45902d48c89c..37970666a453 100644 --- a/sound/i2c/Makefile +++ b/sound/i2c/Makefile @@ -1,15 +1,13 @@ # # Makefile for ALSA -# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz> +# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz> # snd-i2c-objs := i2c.o snd-cs8427-objs := cs8427.o snd-tea6330t-objs := tea6330t.o -ifeq ($(subst m,y,$(CONFIG_L3)),y) - obj-$(CONFIG_L3) += l3/ -endif +obj-$(CONFIG_L3) += l3/ obj-$(CONFIG_SND) += other/ diff --git a/sound/i2c/cs8427.c b/sound/i2c/cs8427.c index 64388cb8d6e5..744366b72345 100644 --- a/sound/i2c/cs8427.c +++ b/sound/i2c/cs8427.c @@ -1,7 +1,7 @@ /* * Routines for control of the CS8427 via i2c bus * IEC958 (S/PDIF) receiver & transmitter by Cirrus Logic - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -32,7 +32,7 @@ static void snd_cs8427_reset(struct snd_i2c_device *cs8427); -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("IEC958 (S/PDIF) receiver & transmitter by Cirrus Logic"); MODULE_LICENSE("GPL"); @@ -229,6 +229,12 @@ int snd_cs8427_create(struct snd_i2c_bus *bus, snd_i2c_lock(bus); err = snd_cs8427_reg_read(device, CS8427_REG_ID_AND_VER); if (err != CS8427_VER8427A) { + /* give second chance */ + snd_printk(KERN_WARNING "invalid CS8427 signature 0x%x: " + "let me try again...\n", err); + err = snd_cs8427_reg_read(device, CS8427_REG_ID_AND_VER); + } + if (err != CS8427_VER8427A) { snd_i2c_unlock(bus); snd_printk(KERN_ERR "unable to find CS8427 signature " "(expected 0x%x, read 0x%x),\n", diff --git a/sound/i2c/i2c.c b/sound/i2c/i2c.c index b60fb1892828..1e58a963b2a7 100644 --- a/sound/i2c/i2c.c +++ b/sound/i2c/i2c.c @@ -2,7 +2,7 @@ * Generic i2c interface for ALSA * * (c) 1998 Gerd Knorr <kraxel@cs.tu-berlin.de> - * Modified for the ALSA driver by Jaroslav Kysela <perex@suse.cz> + * Modified for the ALSA driver by Jaroslav Kysela <perex@perex.cz> * * 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 @@ -28,7 +28,7 @@ #include <sound/core.h> #include <sound/i2c.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Generic i2c interface for ALSA"); MODULE_LICENSE("GPL"); diff --git a/sound/i2c/other/Makefile b/sound/i2c/other/Makefile index 77a8a7c75dd9..703d954238f4 100644 --- a/sound/i2c/other/Makefile +++ b/sound/i2c/other/Makefile @@ -1,6 +1,6 @@ # # Makefile for ALSA -# Copyright (c) 2003 by Jaroslav Kysela <perex@suse.cz> +# Copyright (c) 2003 by Jaroslav Kysela <perex@perex.cz> # snd-ak4114-objs := ak4114.o diff --git a/sound/i2c/other/ak4114.c b/sound/i2c/other/ak4114.c index 1efb973137a6..facde46f957a 100644 --- a/sound/i2c/other/ak4114.c +++ b/sound/i2c/other/ak4114.c @@ -1,7 +1,7 @@ /* * Routines for control of the AK4114 via I2C and 4-wire serial interface * IEC958 (S/PDIF) receiver by Asahi Kasei - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -29,7 +29,7 @@ #include <sound/ak4114.h> #include <sound/asoundef.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("AK4114 IEC958 (S/PDIF) receiver by Asahi Kasei"); MODULE_LICENSE("GPL"); @@ -200,15 +200,7 @@ static int snd_ak4114_in_error_get(struct snd_kcontrol *kcontrol, return 0; } -static int snd_ak4114_in_bit_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_ak4114_in_bit_info snd_ctl_boolean_mono_info static int snd_ak4114_in_bit_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff --git a/sound/i2c/other/ak4117.c b/sound/i2c/other/ak4117.c index c022f29da2f7..ee1585aec99b 100644 --- a/sound/i2c/other/ak4117.c +++ b/sound/i2c/other/ak4117.c @@ -1,7 +1,7 @@ /* * Routines for control of the AK4117 via 4-wire serial interface * IEC958 (S/PDIF) receiver by Asahi Kasei - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -29,7 +29,7 @@ #include <sound/ak4117.h> #include <sound/asoundef.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("AK4117 IEC958 (S/PDIF) receiver by Asahi Kasei"); MODULE_LICENSE("GPL"); @@ -181,15 +181,7 @@ static int snd_ak4117_in_error_get(struct snd_kcontrol *kcontrol, return 0; } -static int snd_ak4117_in_bit_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_ak4117_in_bit_info snd_ctl_boolean_mono_info static int snd_ak4117_in_bit_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c index fd335159f849..de03f689fa2e 100644 --- a/sound/i2c/other/ak4xxx-adda.c +++ b/sound/i2c/other/ak4xxx-adda.c @@ -2,7 +2,7 @@ * ALSA driver for AK4524 / AK4528 / AK4529 / AK4355 / AK4358 / AK4381 * AD and DA converters * - * Copyright (c) 2000-2004 Jaroslav Kysela <perex@suse.cz>, + * Copyright (c) 2000-2004 Jaroslav Kysela <perex@perex.cz>, * Takashi Iwai <tiwai@suse.de> * * This program is free software; you can redistribute it and/or modify @@ -31,7 +31,7 @@ #include <sound/tlv.h> #include <sound/ak4xxx-adda.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>, Takashi Iwai <tiwai@suse.de>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, Takashi Iwai <tiwai@suse.de>"); MODULE_DESCRIPTION("Routines for control of AK452x / AK43xx AD/DA converters"); MODULE_LICENSE("GPL"); @@ -463,15 +463,7 @@ static int snd_akm4xxx_deemphasis_put(struct snd_kcontrol *kcontrol, return change; } -static int ak4xxx_switch_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define ak4xxx_switch_info snd_ctl_boolean_mono_info static int ak4xxx_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff --git a/sound/i2c/other/pt2258.c b/sound/i2c/other/pt2258.c index e91cc3b44de5..00c83d8b32b1 100644 --- a/sound/i2c/other/pt2258.c +++ b/sound/i2c/other/pt2258.c @@ -140,15 +140,7 @@ static int pt2258_stereo_volume_put(struct snd_kcontrol *kcontrol, return -EIO; } -static int pt2258_switch_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define pt2258_switch_info snd_ctl_boolean_mono_info static int pt2258_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c index 4c2fd14c1056..fe31bb5cffb8 100644 --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -1,7 +1,7 @@ /* * ALSA driver for TEA5757/5759 Philips AM/FM radio tuner chips * - * Copyright (c) 2004 Jaroslav Kysela <perex@suse.cz> + * Copyright (c) 2004 Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -28,7 +28,7 @@ #include <sound/core.h> #include <sound/tea575x-tuner.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Routines for control of TEA5757/5759 Philips AM/FM radio tuner chips"); MODULE_LICENSE("GPL"); diff --git a/sound/i2c/tea6330t.c b/sound/i2c/tea6330t.c index ae5b1e3a68ce..9bab744af0ef 100644 --- a/sound/i2c/tea6330t.c +++ b/sound/i2c/tea6330t.c @@ -1,7 +1,7 @@ /* * Routines for control of the TEA6330T circuit via i2c bus * Sound fader control circuit for car radios by Philips Semiconductors - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -27,7 +27,7 @@ #include <sound/control.h> #include <sound/tea6330t.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Routines for control of the TEA6330T circuit via i2c bus"); MODULE_LICENSE("GPL"); @@ -142,15 +142,7 @@ static int snd_tea6330t_put_master_volume(struct snd_kcontrol *kcontrol, .info = snd_tea6330t_info_master_switch, \ .get = snd_tea6330t_get_master_switch, .put = snd_tea6330t_put_master_switch } -static int snd_tea6330t_info_master_switch(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_tea6330t_info_master_switch snd_ctl_boolean_stereo_info static int snd_tea6330t_get_master_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index ea5084abe60f..2639a6ab8f2e 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -191,6 +191,19 @@ config SND_ES18XX To compile this driver as a module, choose M here: the module will be called snd-es18xx. +config SND_SC6000 + tristate "Gallant SC-6000, Audio Excel DSP 16" + depends on SND && HAS_IOPORT + select SND_AD1848_LIB + select SND_OPL3_LIB + select SND_MPU401_UART + help + Say Y here to include support for Gallant SC-6000 card and clones: + Audio Excel DSP 16 and Zoltrix AV302. + + To compile this driver as a module, choose M here: the module + will be called snd-sc6000. + config SND_GUS_SYNTH tristate @@ -414,7 +427,7 @@ config SND_SSCAPE config SND_WAVEFRONT tristate "Turtle Beach Maui,Tropez,Tropez+ (Wavefront)" depends on SND - select FW_LOADER if !SND_WAVEFRONT_FIRMWARE_IN_KERNEL + select FW_LOADER select SND_OPL3_LIB select SND_MPU401_UART select SND_CS4231_LIB @@ -430,8 +443,9 @@ config SND_WAVEFRONT_FIRMWARE_IN_KERNEL depends on SND_WAVEFRONT default y help - Say Y here to include the static firmware built in the kernel - for the Wavefront driver. If you choose N here, you need to - install the firmware files from the alsa-firmware package. + Say Y here to include the static firmware for FX DSP built in + the kernel for the Wavefront driver. If you choose N here, + you need to install the firmware files from the + alsa-firmware package. endmenu diff --git a/sound/isa/Makefile b/sound/isa/Makefile index bb317ccc170f..c0ce7db2a1b5 100644 --- a/sound/isa/Makefile +++ b/sound/isa/Makefile @@ -1,6 +1,6 @@ # # Makefile for ALSA -# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz> +# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz> # snd-adlib-objs := adlib.o @@ -10,6 +10,7 @@ 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 snd-sgalaxy-objs := sgalaxy.o snd-sscape-objs := sscape.o @@ -21,6 +22,7 @@ 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-$(CONFIG_SND_SC6000) += snd-sc6000.o obj-$(CONFIG_SND_SGALAXY) += snd-sgalaxy.o obj-$(CONFIG_SND_SSCAPE) += snd-sscape.o diff --git a/sound/isa/ad1816a/Makefile b/sound/isa/ad1816a/Makefile index 90e00e842e49..487ab23860e3 100644 --- a/sound/isa/ad1816a/Makefile +++ b/sound/isa/ad1816a/Makefile @@ -1,6 +1,6 @@ # # Makefile for ALSA -# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz> +# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz> # snd-ad1816a-objs := ad1816a.o ad1816a_lib.o diff --git a/sound/isa/ad1816a/ad1816a_lib.c b/sound/isa/ad1816a/ad1816a_lib.c index ec9209cd5177..cf18fe4617a1 100644 --- a/sound/isa/ad1816a/ad1816a_lib.c +++ b/sound/isa/ad1816a/ad1816a_lib.c @@ -453,7 +453,6 @@ static int snd_ad1816a_playback_open(struct snd_pcm_substream *substream) if ((error = snd_ad1816a_open(chip, AD1816A_MODE_PLAYBACK)) < 0) return error; - snd_pcm_set_sync(substream); runtime->hw = snd_ad1816a_playback; snd_pcm_limit_isa_dma_size(chip->dma1, &runtime->hw.buffer_bytes_max); snd_pcm_limit_isa_dma_size(chip->dma1, &runtime->hw.period_bytes_max); @@ -469,7 +468,6 @@ static int snd_ad1816a_capture_open(struct snd_pcm_substream *substream) if ((error = snd_ad1816a_open(chip, AD1816A_MODE_CAPTURE)) < 0) return error; - snd_pcm_set_sync(substream); runtime->hw = snd_ad1816a_capture; snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.buffer_bytes_max); snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.period_bytes_max); diff --git a/sound/isa/ad1848/Makefile b/sound/isa/ad1848/Makefile index 45d59998aa69..ae23331e9200 100644 --- a/sound/isa/ad1848/Makefile +++ b/sound/isa/ad1848/Makefile @@ -1,15 +1,12 @@ # # Makefile for ALSA -# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz> +# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz> # snd-ad1848-lib-objs := ad1848_lib.o snd-ad1848-objs := ad1848.o # Toplevel Module Dependency -obj-$(CONFIG_SND_CMI8330) += snd-ad1848-lib.o -obj-$(CONFIG_SND_SGALAXY) += snd-ad1848-lib.o -obj-$(CONFIG_SND_AD1848) += snd-ad1848.o snd-ad1848-lib.o -obj-$(CONFIG_SND_OPTI92X_AD1848) += snd-ad1848-lib.o +obj-$(CONFIG_SND_AD1848) += snd-ad1848.o +obj-$(CONFIG_SND_AD1848_LIB) += snd-ad1848-lib.o -obj-m := $(sort $(obj-m)) diff --git a/sound/isa/ad1848/ad1848.c b/sound/isa/ad1848/ad1848.c index d09a7fa86545..a4710b5e214c 100644 --- a/sound/isa/ad1848/ad1848.c +++ b/sound/isa/ad1848/ad1848.c @@ -1,8 +1,8 @@ /* * Generic driver for AD1848/AD1847/CS4248 chips (0.1 Alpha) * Copyright (c) by Tugrul Galatali <galatalt@stuy.edu>, - * Jaroslav Kysela <perex@suse.cz> - * Based on card-4232.c by Jaroslav Kysela <perex@suse.cz> + * Jaroslav Kysela <perex@perex.cz> + * Based on card-4232.c by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -36,7 +36,7 @@ #define DEV_NAME "ad1848" MODULE_DESCRIPTION(CRD_NAME); -MODULE_AUTHOR("Tugrul Galatali <galatalt@stuy.edu>, Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Tugrul Galatali <galatalt@stuy.edu>, Jaroslav Kysela <perex@perex.cz>"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Analog Devices,AD1848}," "{Analog Devices,AD1847}," diff --git a/sound/isa/ad1848/ad1848_lib.c b/sound/isa/ad1848/ad1848_lib.c index 1bc2e3fd5721..a901cd1ee692 100644 --- a/sound/isa/ad1848/ad1848_lib.c +++ b/sound/isa/ad1848/ad1848_lib.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Routines for control of AD1848/AD1847/CS4248 * * @@ -35,7 +35,7 @@ #include <asm/io.h> #include <asm/dma.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Routines for control of AD1848/AD1847/CS4248"); MODULE_LICENSE("GPL"); @@ -70,7 +70,7 @@ static unsigned int rates[14] = { }; static struct snd_pcm_hw_constraint_list hw_constraints_rates = { - .count = 14, + .count = ARRAY_SIZE(rates), .list = rates, .mask = 0, }; @@ -99,24 +99,32 @@ static unsigned char snd_ad1848_original_image[16] = * Basic I/O functions */ -void snd_ad1848_out(struct snd_ad1848 *chip, - unsigned char reg, - unsigned char value) +static void snd_ad1848_wait(struct snd_ad1848 *chip) { int timeout; - for (timeout = 250; timeout > 0 && (inb(AD1848P(chip, REGSEL)) & AD1848_INIT); timeout--) + for (timeout = 250; timeout > 0; timeout--) { + if ((inb(AD1848P(chip, REGSEL)) & AD1848_INIT) == 0) + break; udelay(100); + } +} + +void snd_ad1848_out(struct snd_ad1848 *chip, + unsigned char reg, + unsigned char value) +{ + snd_ad1848_wait(chip); #ifdef CONFIG_SND_DEBUG if (inb(AD1848P(chip, REGSEL)) & AD1848_INIT) - snd_printk(KERN_WARNING "auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value); + snd_printk(KERN_WARNING "auto calibration time out - " + "reg = 0x%x, value = 0x%x\n", reg, value); #endif outb(chip->mce_bit | reg, AD1848P(chip, REGSEL)); outb(chip->image[reg] = value, AD1848P(chip, REG)); mb(); -#if 0 - printk("codec out - reg 0x%x = 0x%x\n", chip->mce_bit | reg, value); -#endif + snd_printdd("codec out - reg 0x%x = 0x%x\n", + chip->mce_bit | reg, value); } EXPORT_SYMBOL(snd_ad1848_out); @@ -124,10 +132,7 @@ EXPORT_SYMBOL(snd_ad1848_out); static void snd_ad1848_dout(struct snd_ad1848 *chip, unsigned char reg, unsigned char value) { - int timeout; - - for (timeout = 250; timeout > 0 && (inb(AD1848P(chip, REGSEL)) & AD1848_INIT); timeout--) - udelay(100); + snd_ad1848_wait(chip); outb(chip->mce_bit | reg, AD1848P(chip, REGSEL)); outb(value, AD1848P(chip, REG)); mb(); @@ -135,13 +140,11 @@ static void snd_ad1848_dout(struct snd_ad1848 *chip, static unsigned char snd_ad1848_in(struct snd_ad1848 *chip, unsigned char reg) { - int timeout; - - for (timeout = 250; timeout > 0 && (inb(AD1848P(chip, REGSEL)) & AD1848_INIT); timeout--) - udelay(100); + snd_ad1848_wait(chip); #ifdef CONFIG_SND_DEBUG if (inb(AD1848P(chip, REGSEL)) & AD1848_INIT) - snd_printk(KERN_WARNING "auto calibration time out - reg = 0x%x\n", reg); + snd_printk(KERN_WARNING "auto calibration time out - " + "reg = 0x%x\n", reg); #endif outb(chip->mce_bit | reg, AD1848P(chip, REGSEL)); mb(); @@ -183,8 +186,7 @@ static void snd_ad1848_mce_up(struct snd_ad1848 *chip) unsigned long flags; int timeout; - for (timeout = 250; timeout > 0 && (inb(AD1848P(chip, REGSEL)) & AD1848_INIT); timeout--) - udelay(100); + snd_ad1848_wait(chip); #ifdef CONFIG_SND_DEBUG if (inb(AD1848P(chip, REGSEL)) & AD1848_INIT) snd_printk(KERN_WARNING "mce_up - auto calibration time out (0)\n"); @@ -201,9 +203,8 @@ static void snd_ad1848_mce_up(struct snd_ad1848 *chip) static void snd_ad1848_mce_down(struct snd_ad1848 *chip) { - unsigned long flags; - int timeout; - signed long time; + unsigned long flags, timeout; + int reg; spin_lock_irqsave(&chip->reg_lock, flags); for (timeout = 5; timeout > 0; timeout--) @@ -211,61 +212,48 @@ static void snd_ad1848_mce_down(struct snd_ad1848 *chip) /* end of cleanup sequence */ for (timeout = 12000; timeout > 0 && (inb(AD1848P(chip, REGSEL)) & AD1848_INIT); timeout--) udelay(100); -#if 0 - printk("(1) timeout = %i\n", timeout); -#endif + + snd_printdd("(1) timeout = %d\n", timeout); + #ifdef CONFIG_SND_DEBUG if (inb(AD1848P(chip, REGSEL)) & AD1848_INIT) snd_printk(KERN_WARNING "mce_down [0x%lx] - auto calibration time out (0)\n", AD1848P(chip, REGSEL)); #endif + chip->mce_bit &= ~AD1848_MCE; - timeout = inb(AD1848P(chip, REGSEL)); - outb(chip->mce_bit | (timeout & 0x1f), AD1848P(chip, REGSEL)); - if (timeout == 0x80) + reg = inb(AD1848P(chip, REGSEL)); + outb(chip->mce_bit | (reg & 0x1f), AD1848P(chip, REGSEL)); + if (reg == 0x80) snd_printk(KERN_WARNING "mce_down [0x%lx]: serious init problem - codec still busy\n", chip->port); - if ((timeout & AD1848_MCE) == 0) { + if ((reg & AD1848_MCE) == 0) { spin_unlock_irqrestore(&chip->reg_lock, flags); return; } - /* calibration process */ - for (timeout = 500; timeout > 0 && (snd_ad1848_in(chip, AD1848_TEST_INIT) & AD1848_CALIB_IN_PROGRESS) == 0; timeout--); - if ((snd_ad1848_in(chip, AD1848_TEST_INIT) & AD1848_CALIB_IN_PROGRESS) == 0) { - snd_printd("mce_down - auto calibration time out (1)\n"); - spin_unlock_irqrestore(&chip->reg_lock, flags); - return; - } -#if 0 - printk("(2) timeout = %i, jiffies = %li\n", timeout, jiffies); -#endif - time = HZ / 4; - while (snd_ad1848_in(chip, AD1848_TEST_INIT) & AD1848_CALIB_IN_PROGRESS) { + /* + * Wait for auto-calibration (AC) process to finish, i.e. ACI to go low. + * It may take up to 5 sample periods (at most 907 us @ 5.5125 kHz) for + * the process to _start_, so it is important to wait at least that long + * before checking. Otherwise we might think AC has finished when it + * has in fact not begun. It could take 128 (no AC) or 384 (AC) cycles + * for ACI to drop. This gives a wait of at most 70 ms with a more + * typical value of 3-9 ms. + */ + timeout = jiffies + msecs_to_jiffies(250); + do { spin_unlock_irqrestore(&chip->reg_lock, flags); - if (time <= 0) { - snd_printk(KERN_ERR "mce_down - auto calibration time out (2)\n"); - return; - } - time = schedule_timeout(time); + msleep(1); spin_lock_irqsave(&chip->reg_lock, flags); - } -#if 0 - printk("(3) jiffies = %li\n", jiffies); -#endif - time = HZ / 10; - while (inb(AD1848P(chip, REGSEL)) & AD1848_INIT) { - spin_unlock_irqrestore(&chip->reg_lock, flags); - if (time <= 0) { - snd_printk(KERN_ERR "mce_down - auto calibration time out (3)\n"); - return; - } - time = schedule_timeout(time); - spin_lock_irqsave(&chip->reg_lock, flags); - } + reg = snd_ad1848_in(chip, AD1848_TEST_INIT) & + AD1848_CALIB_IN_PROGRESS; + } while (reg && time_before(jiffies, timeout)); spin_unlock_irqrestore(&chip->reg_lock, flags); -#if 0 - printk("(4) jiffies = %li\n", jiffies); - snd_printk("mce_down - exit = 0x%x\n", inb(AD1848P(chip, REGSEL))); -#endif + if (reg) + snd_printk(KERN_ERR + "mce_down - auto calibration time out (2)\n"); + + snd_printdd("(4) jiffies = %lu\n", jiffies); + snd_printd("mce_down - exit = 0x%x\n", inb(AD1848P(chip, REGSEL))); } static unsigned int snd_ad1848_get_count(unsigned char format, @@ -319,11 +307,11 @@ static unsigned char snd_ad1848_get_rate(unsigned int rate) { int i; - for (i = 0; i < 14; i++) + for (i = 0; i < ARRAY_SIZE(rates); i++) if (rate == rates[i]) return freq_bits[i]; snd_BUG(); - return freq_bits[13]; + return freq_bits[ARRAY_SIZE(rates) - 1]; } static int snd_ad1848_ioctl(struct snd_pcm_substream *substream, @@ -390,11 +378,9 @@ static int snd_ad1848_open(struct snd_ad1848 *chip, unsigned int mode) { unsigned long flags; - mutex_lock(&chip->open_mutex); - if (chip->mode & AD1848_MODE_OPEN) { - mutex_unlock(&chip->open_mutex); + if (chip->mode & AD1848_MODE_OPEN) return -EAGAIN; - } + snd_ad1848_mce_down(chip); #ifdef SNDRV_DEBUG_MCE @@ -435,7 +421,6 @@ static int snd_ad1848_open(struct snd_ad1848 *chip, unsigned int mode) spin_unlock_irqrestore(&chip->reg_lock, flags); chip->mode = mode; - mutex_unlock(&chip->open_mutex); return 0; } @@ -444,11 +429,8 @@ static void snd_ad1848_close(struct snd_ad1848 *chip) { unsigned long flags; - mutex_lock(&chip->open_mutex); - if (!chip->mode) { - mutex_unlock(&chip->open_mutex); + if (!chip->mode) return; - } /* disable IRQ */ spin_lock_irqsave(&chip->reg_lock, flags); outb(0, AD1848P(chip, STATUS)); /* clear IRQ */ @@ -474,7 +456,6 @@ static void snd_ad1848_close(struct snd_ad1848 *chip) spin_unlock_irqrestore(&chip->reg_lock, flags); chip->mode = 0; - mutex_unlock(&chip->open_mutex); } /* @@ -892,7 +873,6 @@ int snd_ad1848_create(struct snd_card *card, if (chip == NULL) return -ENOMEM; spin_lock_init(&chip->reg_lock); - mutex_init(&chip->open_mutex); chip->card = card; chip->port = port; chip->irq = -1; diff --git a/sound/isa/cs423x/Makefile b/sound/isa/cs423x/Makefile index 2fb4f7409d7c..5067ee001933 100644 --- a/sound/isa/cs423x/Makefile +++ b/sound/isa/cs423x/Makefile @@ -1,6 +1,6 @@ # # Makefile for ALSA -# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz> +# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz> # snd-cs4231-lib-objs := cs4231_lib.o @@ -10,17 +10,8 @@ snd-cs4232-objs := cs4232.o snd-cs4236-objs := cs4236.o # Toplevel Module Dependency -obj-$(CONFIG_SND_AZT2320) += snd-cs4231-lib.o -obj-$(CONFIG_SND_MIRO) += snd-cs4231-lib.o -obj-$(CONFIG_SND_OPL3SA2) += snd-cs4231-lib.o -obj-$(CONFIG_SND_CS4231) += snd-cs4231.o snd-cs4231-lib.o -obj-$(CONFIG_SND_CS4232) += snd-cs4232.o snd-cs4231-lib.o -obj-$(CONFIG_SND_CS4236) += snd-cs4236.o snd-cs4236-lib.o snd-cs4231-lib.o -obj-$(CONFIG_SND_GUSMAX) += snd-cs4231-lib.o -obj-$(CONFIG_SND_INTERWAVE) += snd-cs4231-lib.o -obj-$(CONFIG_SND_INTERWAVE_STB) += snd-cs4231-lib.o -obj-$(CONFIG_SND_OPTI92X_CS4231) += snd-cs4231-lib.o -obj-$(CONFIG_SND_WAVEFRONT) += snd-cs4231-lib.o -obj-$(CONFIG_SND_SSCAPE) += snd-cs4231-lib.o +obj-$(CONFIG_SND_CS4231_LIB) += snd-cs4231-lib.o +obj-$(CONFIG_SND_CS4231) += snd-cs4231.o +obj-$(CONFIG_SND_CS4232) += snd-cs4232.o +obj-$(CONFIG_SND_CS4236) += snd-cs4236.o snd-cs4236-lib.o -obj-m := $(sort $(obj-m)) diff --git a/sound/isa/cs423x/cs4231.c b/sound/isa/cs423x/cs4231.c index ac4041134150..13db6842eaaa 100644 --- a/sound/isa/cs423x/cs4231.c +++ b/sound/isa/cs423x/cs4231.c @@ -1,6 +1,6 @@ /* * Generic driver for CS4231 chips - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Originally the CS4232/CS4232A driver, modified for use on CS4231 by * Tugrul Galatali <galatalt@stuy.edu> * @@ -36,7 +36,7 @@ #define DEV_NAME "cs4231" MODULE_DESCRIPTION(CRD_NAME); -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Crystal Semiconductors,CS4231}}"); diff --git a/sound/isa/cs423x/cs4231_lib.c b/sound/isa/cs423x/cs4231_lib.c index 914d77b61b0c..a5eb9659b519 100644 --- a/sound/isa/cs423x/cs4231_lib.c +++ b/sound/isa/cs423x/cs4231_lib.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Routines for control of CS4231(A)/CS4232/InterWave & compatible chips * * Bugs: @@ -39,7 +39,7 @@ #include <asm/dma.h> #include <asm/irq.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Routines for control of CS4231(A)/CS4232/InterWave & compatible chips"); MODULE_LICENSE("GPL"); @@ -74,7 +74,7 @@ static unsigned int rates[14] = { }; static struct snd_pcm_hw_constraint_list hw_constraints_rates = { - .count = 14, + .count = ARRAY_SIZE(rates), .list = rates, .mask = 0, }; @@ -134,29 +134,31 @@ static inline u8 cs4231_inb(struct snd_cs4231 *chip, u8 offset) return inb(chip->port + offset); } -static void snd_cs4231_outm(struct snd_cs4231 *chip, unsigned char reg, - unsigned char mask, unsigned char value) +static void snd_cs4231_wait(struct snd_cs4231 *chip) { int timeout; - unsigned char tmp; for (timeout = 250; timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); timeout--) udelay(100); +} + +static void snd_cs4231_outm(struct snd_cs4231 *chip, unsigned char reg, + unsigned char mask, unsigned char value) +{ + unsigned char tmp = (chip->image[reg] & mask) | value; + + snd_cs4231_wait(chip); #ifdef CONFIG_SND_DEBUG if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) snd_printk("outm: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value); #endif - if (chip->calibrate_mute) { - chip->image[reg] &= mask; - chip->image[reg] |= value; - } else { + chip->image[reg] = tmp; + if (!chip->calibrate_mute) { cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); - mb(); - tmp = (chip->image[reg] & mask) | value; + wmb(); cs4231_outb(chip, CS4231P(REG), tmp); - chip->image[reg] = tmp; mb(); } } @@ -176,12 +178,7 @@ static void snd_cs4231_dout(struct snd_cs4231 *chip, unsigned char reg, unsigned void snd_cs4231_out(struct snd_cs4231 *chip, unsigned char reg, unsigned char value) { - int timeout; - - for (timeout = 250; - timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); - timeout--) - udelay(100); + snd_cs4231_wait(chip); #ifdef CONFIG_SND_DEBUG if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) snd_printk("out: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value); @@ -190,19 +187,13 @@ void snd_cs4231_out(struct snd_cs4231 *chip, unsigned char reg, unsigned char va cs4231_outb(chip, CS4231P(REG), value); chip->image[reg] = value; mb(); -#if 0 - printk("codec out - reg 0x%x = 0x%x\n", chip->mce_bit | reg, value); -#endif + snd_printdd("codec out - reg 0x%x = 0x%x\n", + chip->mce_bit | reg, value); } unsigned char snd_cs4231_in(struct snd_cs4231 *chip, unsigned char reg) { - int timeout; - - for (timeout = 250; - timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); - timeout--) - udelay(100); + snd_cs4231_wait(chip); #ifdef CONFIG_SND_DEBUG if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) snd_printk("in: auto calibration time out - reg = 0x%x\n", reg); @@ -304,8 +295,7 @@ void snd_cs4231_mce_up(struct snd_cs4231 *chip) unsigned long flags; int timeout; - for (timeout = 250; timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); timeout--) - udelay(100); + snd_cs4231_wait(chip); #ifdef CONFIG_SND_DEBUG if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) snd_printk("mce_up - auto calibration time out (0)\n"); @@ -323,12 +313,11 @@ void snd_cs4231_mce_up(struct snd_cs4231 *chip) void snd_cs4231_mce_down(struct snd_cs4231 *chip) { unsigned long flags; + unsigned long end_time; int timeout; snd_cs4231_busy_wait(chip); -#if 0 - printk("(1) timeout = %i\n", timeout); -#endif + #ifdef CONFIG_SND_DEBUG if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) snd_printk("mce_down [0x%lx] - auto calibration time out (0)\n", (long)CS4231P(REGSEL)); @@ -346,42 +335,42 @@ void snd_cs4231_mce_down(struct snd_cs4231 *chip) } snd_cs4231_busy_wait(chip); - /* calibration process */ + /* + * Wait for (possible -- during init auto-calibration may not be set) + * calibration process to start. Needs upto 5 sample periods on AD1848 + * which at the slowest possible rate of 5.5125 kHz means 907 us. + */ + msleep(1); - for (timeout = 500; timeout > 0 && (snd_cs4231_in(chip, CS4231_TEST_INIT) & CS4231_CALIB_IN_PROGRESS) == 0; timeout--) - udelay(10); - if ((snd_cs4231_in(chip, CS4231_TEST_INIT) & CS4231_CALIB_IN_PROGRESS) == 0) { - snd_printd("cs4231_mce_down - auto calibration time out (1)\n"); - return; - } -#if 0 - printk("(2) timeout = %i, jiffies = %li\n", timeout, jiffies); -#endif - /* in 10 ms increments, check condition, up to 250 ms */ - timeout = 25; - while (snd_cs4231_in(chip, CS4231_TEST_INIT) & CS4231_CALIB_IN_PROGRESS) { - if (--timeout < 0) { - snd_printk("mce_down - auto calibration time out (2)\n"); + snd_printdd("(1) jiffies = %lu\n", jiffies); + + /* check condition up to 250 ms */ + end_time = jiffies + msecs_to_jiffies(250); + while (snd_cs4231_in(chip, CS4231_TEST_INIT) & + CS4231_CALIB_IN_PROGRESS) { + + if (time_after(jiffies, end_time)) { + snd_printk(KERN_ERR "mce_down - " + "auto calibration time out (2)\n"); return; } - msleep(10); + msleep(1); } -#if 0 - printk("(3) jiffies = %li\n", jiffies); -#endif - /* in 10 ms increments, check condition, up to 100 ms */ - timeout = 10; + + snd_printdd("(2) jiffies = %lu\n", jiffies); + + /* check condition up to 100 ms */ + end_time = jiffies + msecs_to_jiffies(100); while (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) { - if (--timeout < 0) { + if (time_after(jiffies, end_time)) { snd_printk(KERN_ERR "mce_down - auto calibration time out (3)\n"); return; } - msleep(10); + msleep(1); } -#if 0 - printk("(4) jiffies = %li\n", jiffies); - snd_printk("mce_down - exit = 0x%x\n", cs4231_inb(chip, CS4231P(REGSEL))); -#endif + + snd_printdd("(3) jiffies = %lu\n", jiffies); + snd_printd("mce_down - exit = 0x%x\n", cs4231_inb(chip, CS4231P(REGSEL))); } static unsigned int snd_cs4231_get_count(unsigned char format, unsigned int size) @@ -459,11 +448,11 @@ static unsigned char snd_cs4231_get_rate(unsigned int rate) { int i; - for (i = 0; i < 14; i++) + for (i = 0; i < ARRAY_SIZE(rates); i++) if (rate == rates[i]) return freq_bits[i]; // snd_BUG(); - return freq_bits[13]; + return freq_bits[ARRAY_SIZE(rates) - 1]; } static unsigned char snd_cs4231_get_format(struct snd_cs4231 *chip, @@ -555,6 +544,8 @@ static void snd_cs4231_playback_format(struct snd_cs4231 *chip, snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, chip->image[CS4231_PLAYBK_FORMAT] = pdfr); } spin_unlock_irqrestore(&chip->reg_lock, flags); + if (chip->hardware == CS4231_HW_OPL3SA2) + udelay(100); /* this seems to help */ snd_cs4231_mce_down(chip); } snd_cs4231_calibrate_mute(chip, 0); diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index 1a14f33b6ab0..5784b43f4123 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -1,6 +1,6 @@ /* * Driver for generic CS4232/CS4235/CS4236/CS4236B/CS4237B/CS4238B/CS4239 chips - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -32,7 +32,7 @@ #include <sound/opl3.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_LICENSE("GPL"); #ifdef CS4232 MODULE_DESCRIPTION("Cirrus Logic CS4232"); diff --git a/sound/isa/cs423x/cs4236_lib.c b/sound/isa/cs423x/cs4236_lib.c index 7a5a6c71f5e4..6bd064470d4c 100644 --- a/sound/isa/cs423x/cs4236_lib.c +++ b/sound/isa/cs423x/cs4236_lib.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Routines for control of CS4235/4236B/4237B/4238B/4239 chips * * Note: @@ -89,7 +89,7 @@ #include <sound/cs4231.h> #include <sound/asoundef.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Routines for control of CS4235/4236B/4237B/4238B/4239 chips"); MODULE_LICENSE("GPL"); diff --git a/sound/isa/es1688/Makefile b/sound/isa/es1688/Makefile index 501c8bf903af..aee1e4ddb22a 100644 --- a/sound/isa/es1688/Makefile +++ b/sound/isa/es1688/Makefile @@ -1,6 +1,6 @@ # # Makefile for ALSA -# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz> +# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz> # snd-es1688-lib-objs := es1688_lib.o diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index edc398712e8b..74bbc92f2e7c 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -1,6 +1,6 @@ /* * Driver for generic ESS AudioDrive ESx688 soundcards - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -39,7 +39,7 @@ #define DEV_NAME "es1688" MODULE_DESCRIPTION(CRD_NAME); -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{ESS,ES688 PnP AudioDrive,pnp:ESS0100}," "{ESS,ES1688 PnP AudioDrive,pnp:ESS0102}," diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c index a2ab99f2ac35..5c26d495daa8 100644 --- a/sound/isa/es1688/es1688_lib.c +++ b/sound/isa/es1688/es1688_lib.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Routines for control of ESS ES1688/688/488 chip * * @@ -32,7 +32,7 @@ #include <asm/io.h> #include <asm/dma.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("ESS ESx688 lowlevel module"); MODULE_LICENSE("GPL"); diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index f7732bf90be3..4a7367a8ff9d 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -1071,14 +1071,7 @@ static int snd_es18xx_put_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem return (snd_es18xx_mixer_bits(chip, 0x1c, 0x07, val) != val) || retVal; } -static int snd_es18xx_info_spatializer_enable(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_es18xx_info_spatializer_enable snd_ctl_boolean_mono_info static int snd_es18xx_get_spatializer_enable(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1120,14 +1113,7 @@ static int snd_es18xx_get_hw_volume(struct snd_kcontrol *kcontrol, struct snd_ct return 0; } -static int snd_es18xx_info_hw_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_es18xx_info_hw_switch snd_ctl_boolean_stereo_info static int snd_es18xx_get_hw_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -2042,6 +2028,7 @@ static int pnpc_registered; static struct pnp_device_id snd_audiodrive_pnpbiosids[] = { { .id = "ESS1869" }, + { .id = "ESS1879" }, { .id = "" } /* end */ }; diff --git a/sound/isa/gus/Makefile b/sound/isa/gus/Makefile index bae5dbd6c8e5..df3d59f25f5e 100644 --- a/sound/isa/gus/Makefile +++ b/sound/isa/gus/Makefile @@ -1,6 +1,6 @@ # # Makefile for ALSA -# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz> +# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz> # snd-gus-lib-objs := gus_main.o \ diff --git a/sound/isa/gus/gus_dma.c b/sound/isa/gus/gus_dma.c index 44ee5d3674a1..fc905141e8a5 100644 --- a/sound/isa/gus/gus_dma.c +++ b/sound/isa/gus/gus_dma.c @@ -1,6 +1,6 @@ /* * Routines for GF1 DMA control - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/sound/isa/gus/gus_dram.c b/sound/isa/gus/gus_dram.c index f22fe7967fcc..9eaa932f6efe 100644 --- a/sound/isa/gus/gus_dram.c +++ b/sound/isa/gus/gus_dram.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * DRAM access routines * * diff --git a/sound/isa/gus/gus_instr.c b/sound/isa/gus/gus_instr.c index d0c38e1856ef..bf137ea72329 100644 --- a/sound/isa/gus/gus_instr.c +++ b/sound/isa/gus/gus_instr.c @@ -1,6 +1,6 @@ /* * Routines for Gravis UltraSound soundcards - Synthesizer - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/sound/isa/gus/gus_io.c b/sound/isa/gus/gus_io.c index 9b1fe292de4d..3d4f899285ef 100644 --- a/sound/isa/gus/gus_io.c +++ b/sound/isa/gus/gus_io.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * I/O routines for GF1/InterWave synthesizer chips * * diff --git a/sound/isa/gus/gus_irq.c b/sound/isa/gus/gus_irq.c index 537d3cfe41f3..cd9a6f1c99e6 100644 --- a/sound/isa/gus/gus_irq.c +++ b/sound/isa/gus/gus_irq.c @@ -1,6 +1,6 @@ /* * Routine for IRQ handling from GF1/InterWave chip - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -45,11 +45,13 @@ __again: // snd_printk("IRQ: status = 0x%x\n", status); if (status & 0x02) { STAT_ADD(gus->gf1.interrupt_stat_midi_in); - gus->gf1.interrupt_handler_midi_in(gus); + if (gus->gf1.interrupt_handler_midi_in) + gus->gf1.interrupt_handler_midi_in(gus); } if (status & 0x01) { STAT_ADD(gus->gf1.interrupt_stat_midi_out); - gus->gf1.interrupt_handler_midi_out(gus); + if (gus->gf1.interrupt_handler_midi_out) + gus->gf1.interrupt_handler_midi_out(gus); } if (status & (0x20 | 0x40)) { unsigned int already, _current_; @@ -85,20 +87,24 @@ __again: } if (status & 0x04) { STAT_ADD(gus->gf1.interrupt_stat_timer1); - gus->gf1.interrupt_handler_timer1(gus); + if (gus->gf1.interrupt_handler_timer1) + gus->gf1.interrupt_handler_timer1(gus); } if (status & 0x08) { STAT_ADD(gus->gf1.interrupt_stat_timer2); - gus->gf1.interrupt_handler_timer2(gus); + if (gus->gf1.interrupt_handler_timer2) + gus->gf1.interrupt_handler_timer2(gus); } if (status & 0x80) { if (snd_gf1_i_look8(gus, SNDRV_GF1_GB_DRAM_DMA_CONTROL) & 0x40) { STAT_ADD(gus->gf1.interrupt_stat_dma_write); - gus->gf1.interrupt_handler_dma_write(gus); + if (gus->gf1.interrupt_handler_dma_write) + gus->gf1.interrupt_handler_dma_write(gus); } if (snd_gf1_i_look8(gus, SNDRV_GF1_GB_REC_DMA_CONTROL) & 0x40) { STAT_ADD(gus->gf1.interrupt_stat_dma_read); - gus->gf1.interrupt_handler_dma_read(gus); + if (gus->gf1.interrupt_handler_dma_read) + gus->gf1.interrupt_handler_dma_read(gus); } } if (--loop > 0) diff --git a/sound/isa/gus/gus_main.c b/sound/isa/gus/gus_main.c index 8ced5e81b9a7..b14d5d6d9a32 100644 --- a/sound/isa/gus/gus_main.c +++ b/sound/isa/gus/gus_main.c @@ -1,6 +1,6 @@ /* * Routines for Gravis UltraSound soundcards - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -31,7 +31,7 @@ #include <asm/dma.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Routines for Gravis UltraSound soundcards"); MODULE_LICENSE("GPL"); @@ -154,6 +154,14 @@ int snd_gus_create(struct snd_card *card, gus = kzalloc(sizeof(*gus), GFP_KERNEL); if (gus == NULL) return -ENOMEM; + spin_lock_init(&gus->reg_lock); + spin_lock_init(&gus->voice_alloc); + spin_lock_init(&gus->active_voice_lock); + spin_lock_init(&gus->event_lock); + spin_lock_init(&gus->dma_lock); + spin_lock_init(&gus->pcm_volume_level_lock); + spin_lock_init(&gus->uart_cmd_lock); + mutex_init(&gus->dma_mutex); gus->gf1.irq = -1; gus->gf1.dma1 = -1; gus->gf1.dma2 = -1; @@ -218,14 +226,6 @@ int snd_gus_create(struct snd_card *card, gus->gf1.pcm_channels = pcm_channels; gus->gf1.volume_ramp = 25; gus->gf1.smooth_pan = 1; - spin_lock_init(&gus->reg_lock); - spin_lock_init(&gus->voice_alloc); - spin_lock_init(&gus->active_voice_lock); - spin_lock_init(&gus->event_lock); - spin_lock_init(&gus->dma_lock); - spin_lock_init(&gus->pcm_volume_level_lock); - spin_lock_init(&gus->uart_cmd_lock); - mutex_init(&gus->dma_mutex); if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, gus, &ops)) < 0) { snd_gus_free(gus); return err; @@ -398,7 +398,7 @@ static int snd_gus_check_version(struct snd_gus_card * gus) gus->ess_flag = 1; } else { snd_printk(KERN_ERR "unknown GF1 revision number at 0x%lx - 0x%x (0x%x)\n", gus->gf1.port, rev, val); - snd_printk(KERN_ERR " please - report to <perex@suse.cz>\n"); + snd_printk(KERN_ERR " please - report to <perex@perex.cz>\n"); } } } diff --git a/sound/isa/gus/gus_mem.c b/sound/isa/gus/gus_mem.c index 7107753b85b5..bcf4656853c4 100644 --- a/sound/isa/gus/gus_mem.c +++ b/sound/isa/gus/gus_mem.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * GUS's memory allocation routines / bottom layer * * diff --git a/sound/isa/gus/gus_mem_proc.c b/sound/isa/gus/gus_mem_proc.c index 80f0a83818b2..f69a44728ebf 100644 --- a/sound/isa/gus/gus_mem_proc.c +++ b/sound/isa/gus/gus_mem_proc.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * GUS's memory access via proc filesystem * * diff --git a/sound/isa/gus/gus_mixer.c b/sound/isa/gus/gus_mixer.c index acc25a297200..a96253e16654 100644 --- a/sound/isa/gus/gus_mixer.c +++ b/sound/isa/gus/gus_mixer.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Routines for control of ICS 2101 chip and "mixer" in GF1 chip * * @@ -36,14 +36,7 @@ .get = snd_gf1_get_single, .put = snd_gf1_put_single, \ .private_value = shift | (invert << 8) } -static int snd_gf1_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_gf1_info_single snd_ctl_boolean_mono_info static int snd_gf1_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { diff --git a/sound/isa/gus/gus_pcm.c b/sound/isa/gus/gus_pcm.c index c7f95e7aa018..a7971f5ffe63 100644 --- a/sound/isa/gus/gus_pcm.c +++ b/sound/isa/gus/gus_pcm.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Routines for control of GF1 chip (PCM things) * * InterWave chips supports interleaved DMA, but this feature isn't used in diff --git a/sound/isa/gus/gus_reset.c b/sound/isa/gus/gus_reset.c index b263655c4116..20cfdb87f84a 100644 --- a/sound/isa/gus/gus_reset.c +++ b/sound/isa/gus/gus_reset.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/sound/isa/gus/gus_sample.c b/sound/isa/gus/gus_sample.c index 9e0c55ab25b2..cba0829a7106 100644 --- a/sound/isa/gus/gus_sample.c +++ b/sound/isa/gus/gus_sample.c @@ -1,6 +1,6 @@ /* * Routines for Gravis UltraSound soundcards - Sample support - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/sound/isa/gus/gus_simple.c b/sound/isa/gus/gus_simple.c index dcad6ed0198c..39d121e2c8c4 100644 --- a/sound/isa/gus/gus_simple.c +++ b/sound/isa/gus/gus_simple.c @@ -1,6 +1,6 @@ /* * Routines for Gravis UltraSound soundcards - Simple instrument handlers - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/sound/isa/gus/gus_synth.c b/sound/isa/gus/gus_synth.c index 3e4d4d6edd8b..2c2051782aa2 100644 --- a/sound/isa/gus/gus_synth.c +++ b/sound/isa/gus/gus_synth.c @@ -1,6 +1,6 @@ /* * Routines for Gravis UltraSound soundcards - Synthesizer - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -26,7 +26,7 @@ #include <sound/gus.h> #include <sound/seq_device.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Routines for Gravis UltraSound soundcards - Synthesizer"); MODULE_LICENSE("GPL"); diff --git a/sound/isa/gus/gus_tables.h b/sound/isa/gus/gus_tables.h index 4adf098d3269..42a4ca0d622b 100644 --- a/sound/isa/gus/gus_tables.h +++ b/sound/isa/gus/gus_tables.h @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/sound/isa/gus/gus_timer.c b/sound/isa/gus/gus_timer.c index a43b662f17c7..99eac573c414 100644 --- a/sound/isa/gus/gus_timer.c +++ b/sound/isa/gus/gus_timer.c @@ -1,6 +1,6 @@ /* * Routines for Gravis UltraSound soundcards - Timers - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * GUS have similar timers as AdLib (OPL2/OPL3 chips). * diff --git a/sound/isa/gus/gus_uart.c b/sound/isa/gus/gus_uart.c index 654290a8b21c..e6fd9b01c492 100644 --- a/sound/isa/gus/gus_uart.c +++ b/sound/isa/gus/gus_uart.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Routines for the GF1 MIDI interface - like UART 6850 * * diff --git a/sound/isa/gus/gus_volume.c b/sound/isa/gus/gus_volume.c index dbbc0a6d7659..71a67744a14b 100644 --- a/sound/isa/gus/gus_volume.c +++ b/sound/isa/gus/gus_volume.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/sound/isa/gus/gusclassic.c b/sound/isa/gus/gusclassic.c index 8f23f433d491..29e422b00b58 100644 --- a/sound/isa/gus/gusclassic.c +++ b/sound/isa/gus/gusclassic.c @@ -1,6 +1,6 @@ /* * Driver for Gravis UltraSound Classic soundcard - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -37,7 +37,7 @@ #define DEV_NAME "gusclassic" MODULE_DESCRIPTION(CRD_NAME); -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Gravis,UltraSound Classic}}"); diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c index 0aeaa6cf6cf0..fc59536c918e 100644 --- a/sound/isa/gus/gusextreme.c +++ b/sound/isa/gus/gusextreme.c @@ -1,6 +1,6 @@ /* * Driver for Gravis UltraSound Extreme soundcards - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -41,7 +41,7 @@ #define DEV_NAME "gusextreme" MODULE_DESCRIPTION(CRD_NAME); -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Gravis,UltraSound Extreme}}"); diff --git a/sound/isa/gus/gusmax.c b/sound/isa/gus/gusmax.c index 708783d4351f..4922f5da08f9 100644 --- a/sound/isa/gus/gusmax.c +++ b/sound/isa/gus/gusmax.c @@ -1,6 +1,6 @@ /* * Driver for Gravis UltraSound MAX soundcard - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -34,7 +34,7 @@ #define SNDRV_LEGACY_FIND_FREE_DMA #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Gravis UltraSound MAX"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Gravis,UltraSound MAX}}"); diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index 0220cdbe1a2a..2091c50b2e3e 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -1,6 +1,6 @@ /* * Driver for AMD InterWave soundcard - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -41,7 +41,7 @@ #define SNDRV_LEGACY_FIND_FREE_DMA #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_LICENSE("GPL"); #ifndef SNDRV_STB MODULE_DESCRIPTION("AMD InterWave"); diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index e70db32991d9..59af9ab7191f 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -1,6 +1,6 @@ /* * Driver for Yamaha OPL3-SA[2,3] soundcards - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -37,7 +37,7 @@ #include <asm/io.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Yamaha OPL3SA2+"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Yamaha,YMF719E-S}," @@ -253,6 +253,7 @@ static int __devinit snd_opl3sa2_detect(struct snd_opl3sa2 *chip) /* 0x03 - YM715B */ /* 0x04 - YM719 - OPL-SA4? */ /* 0x05 - OPL3-SA3 - Libretto 100 */ + /* 0x07 - unknown - Neomagic MagicWave 3D */ break; } str[0] = chip->version + '0'; diff --git a/sound/isa/opti9xx/Makefile b/sound/isa/opti9xx/Makefile index 0e41bfd5a403..b4d894db257a 100644 --- a/sound/isa/opti9xx/Makefile +++ b/sound/isa/opti9xx/Makefile @@ -1,6 +1,6 @@ # # Makefile for ALSA -# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz> +# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz> # snd-opti92x-ad1848-objs := opti92x-ad1848.o diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index cd29b30b362e..d295936611f8 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -242,14 +242,7 @@ static int aci_setvalue(struct snd_miro * miro, unsigned char index, int value) * MIXER part */ -static int snd_miro_info_capture(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - - return 0; -} +#define snd_miro_info_capture snd_ctl_boolean_mono_info static int snd_miro_get_capture(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -344,14 +337,7 @@ static int snd_miro_put_preamp(struct snd_kcontrol *kcontrol, return change; } -static int snd_miro_info_amp(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - - return 0; -} +#define snd_miro_info_amp snd_ctl_boolean_mono_info static int snd_miro_get_amp(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 049d479ce2b3..ee1a824d8fc0 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -501,6 +501,16 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip) (chip->hardware == OPTi9XX_HW_82C930 ? 0x00 : 0x04), 0x34); snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(5), 0x20, 0xbf); + /* + * The BTC 1817DW has QS1000 wavetable which is connected + * to the serial digital input of the OPTI931. + */ + snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(21), 0x82, 0xff); + /* + * This bit sets OPTI931 to automaticaly select FM + * or digital input signal. + */ + snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(26), 0x01, 0x01); break; #endif /* OPTi93X */ @@ -1732,11 +1742,11 @@ 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); + fm_port = pnp_port_start(pdev, 1) + 8; #else if (pid->driver_data != 0x0924) port = pnp_port_start(pdev, 1); - fm_port = pnp_port_start(pdev, 2); + fm_port = pnp_port_start(pdev, 2) + 8; #endif /* OPTi93X */ irq = pnp_irq(pdev, 0); dma1 = pnp_dma(pdev, 0); diff --git a/sound/isa/sb/Makefile b/sound/isa/sb/Makefile index 556e66928029..c9d1c986d70e 100644 --- a/sound/isa/sb/Makefile +++ b/sound/isa/sb/Makefile @@ -1,6 +1,6 @@ # # Makefile for ALSA -# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz> +# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz> # snd-sb-common-objs := sb_common.o sb_mixer.o diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c index 658179e86142..4eea84cfd4f4 100644 --- a/sound/isa/sb/emu8000.c +++ b/sound/isa/sb/emu8000.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * and (c) 1999 Steve Ratcliffe <steve@parabola.demon.co.uk> * Copyright (C) 1999-2000 Takashi Iwai <tiwai@suse.de> * diff --git a/sound/isa/sb/emu8000_synth.c b/sound/isa/sb/emu8000_synth.c index 3d72742b342f..0c7905c85b76 100644 --- a/sound/isa/sb/emu8000_synth.c +++ b/sound/isa/sb/emu8000_synth.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * and (c) 1999 Steve Ratcliffe <steve@parabola.demon.co.uk> * Copyright (C) 1999-2000 Takashi Iwai <tiwai@suse.de> * diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c index c4ba24bfd27c..e7f9edd92626 100644 --- a/sound/isa/sb/sb16.c +++ b/sound/isa/sb/sb16.c @@ -1,6 +1,6 @@ /* * Driver for SoundBlaster 16/AWE32/AWE64 soundcards - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -44,7 +44,7 @@ #define PFX "sb16: " #endif -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_LICENSE("GPL"); #ifndef SNDRV_SBAWE MODULE_DESCRIPTION("Sound Blaster 16"); diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c index b279f2308aef..3682059787ab 100644 --- a/sound/isa/sb/sb16_csp.c +++ b/sound/isa/sb/sb16_csp.c @@ -979,14 +979,7 @@ static int snd_sb_csp_restart(struct snd_sb_csp * p) * QSound mixer control for PCM */ -static int snd_sb_qsound_switch_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_sb_qsound_switch_info snd_ctl_boolean_mono_info static int snd_sb_qsound_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { diff --git a/sound/isa/sb/sb16_main.c b/sound/isa/sb/sb16_main.c index 5d4d3aafe2d5..c06754f7ee5d 100644 --- a/sound/isa/sb/sb16_main.c +++ b/sound/isa/sb/sb16_main.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Routines for control of 16-bit SoundBlaster cards and clones * Note: This is very ugly hardware which uses one 8-bit DMA channel and * second 16-bit DMA channel. Unfortunately 8-bit DMA channel can't @@ -45,7 +45,7 @@ #include <sound/control.h> #include <sound/info.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Routines for control of 16-bit SoundBlaster cards and clones"); MODULE_LICENSE("GPL"); diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c index a1b3786b391e..f933aef7d8a9 100644 --- a/sound/isa/sb/sb8.c +++ b/sound/isa/sb/sb8.c @@ -1,6 +1,6 @@ /* * Driver for SoundBlaster 1.0/2.0/Pro soundcards and compatible - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -31,7 +31,7 @@ #include <sound/opl3.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Sound Blaster 1.0/2.0/Pro"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Creative Labs,SB 1.0/SB 2.0/SB Pro}}"); diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index aea9e5ec7b36..bee894b3f5c7 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Uros Bizjak <uros@kss-loka.si> * * Routines for control of 8-bit SoundBlaster cards and clones @@ -38,7 +38,7 @@ #include <sound/core.h> #include <sound/sb.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>, Uros Bizjak <uros@kss-loka.si>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, Uros Bizjak <uros@kss-loka.si>"); MODULE_DESCRIPTION("Routines for control of 8-bit SoundBlaster cards and clones"); MODULE_LICENSE("GPL"); diff --git a/sound/isa/sb/sb8_midi.c b/sound/isa/sb/sb8_midi.c index 0b67edd7ac6e..e56e5633411c 100644 --- a/sound/isa/sb/sb8_midi.c +++ b/sound/isa/sb/sb8_midi.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Routines for control of SoundBlaster cards - MIDI interface * * This program is free software; you can redistribute it and/or modify diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c index efa9d5c2558a..176193c05101 100644 --- a/sound/isa/sb/sb_common.c +++ b/sound/isa/sb/sb_common.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Uros Bizjak <uros@kss-loka.si> * * Lowlevel routines for control of Sound Blaster cards @@ -33,7 +33,7 @@ #include <asm/io.h> #include <asm/dma.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("ALSA lowlevel driver for Sound Blaster cards"); MODULE_LICENSE("GPL"); @@ -234,7 +234,9 @@ int snd_sbdsp_create(struct snd_card *card, chip->dma16 = -1; chip->port = port; - if (request_irq(irq, irq_handler, hardware == SB_HW_ALS4000 ? + if (request_irq(irq, irq_handler, + (hardware == SB_HW_ALS4000 || + hardware == SB_HW_CS5530) ? IRQF_SHARED : IRQF_DISABLED, "SoundBlaster", (void *) chip)) { snd_printk(KERN_ERR "sb: can't grab irq %d\n", irq); diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c index 3d4befcff28e..03241cd5aaef 100644 --- a/sound/isa/sb/sb_mixer.c +++ b/sound/isa/sb/sb_mixer.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Routines for Sound Blaster mixer control * * diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c new file mode 100644 index 000000000000..94daf8399994 --- /dev/null +++ b/sound/isa/sc6000.c @@ -0,0 +1,656 @@ +/* + * Driver for Gallant SC-6000 soundcard. This card is also known as + * Audio Excel DSP 16 or Zoltrix AV302. + * These cards use CompuMedia ASC-9308 chip + AD1848 codec. + * + * Copyright (C) 2007 Krzysztof Helt <krzysztof.h1@wp.pl> + * + * I don't have documentation for this card. I used the driver + * for OSS/Free included in the kernel source as reference. + * + * 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 <sound/driver.h> +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/isa.h> +#include <linux/io.h> +#include <asm/dma.h> +#include <sound/core.h> +#include <sound/ad1848.h> +#include <sound/opl3.h> +#include <sound/mpu401.h> +#include <sound/control.h> +#define SNDRV_LEGACY_FIND_FREE_IRQ +#define SNDRV_LEGACY_FIND_FREE_DMA +#include <sound/initval.h> + +MODULE_AUTHOR("Krzysztof Helt"); +MODULE_DESCRIPTION("Gallant SC-6000"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{Gallant, SC-6000}," + "{AudioExcel, Audio Excel DSP 16}," + "{Zoltrix, AV302}}"); + +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; /* 0x220, 0x240 */ +static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5, 7, 9, 10, 11 */ +static long mss_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x530, 0xe80 */ +static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; + /* 0x300, 0x310, 0x320, 0x330 */ +static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5, 7, 9, 10, 0 */ +static int dma[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0, 1, 3 */ + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for sc-6000 based soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for sc-6000 based soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable sc-6000 based soundcard."); +module_param_array(port, long, NULL, 0444); +MODULE_PARM_DESC(port, "Port # for sc-6000 driver."); +module_param_array(mss_port, long, NULL, 0444); +MODULE_PARM_DESC(mss_port, "MSS Port # for sc-6000 driver."); +module_param_array(mpu_port, long, NULL, 0444); +MODULE_PARM_DESC(mpu_port, "MPU-401 port # for sc-6000 driver."); +module_param_array(irq, int, NULL, 0444); +MODULE_PARM_DESC(irq, "IRQ # for sc-6000 driver."); +module_param_array(mpu_irq, int, NULL, 0444); +MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for sc-6000 driver."); +module_param_array(dma, int, NULL, 0444); +MODULE_PARM_DESC(dma, "DMA # for sc-6000 driver."); + +/* + * Commands of SC6000's DSP (SBPRO+special). + * Some of them are COMMAND_xx, in the future they may change. + */ +#define WRITE_MDIRQ_CFG 0x50 /* Set M&I&DRQ mask (the real config) */ +#define COMMAND_52 0x52 /* */ +#define READ_HARD_CFG 0x58 /* Read Hardware Config (I/O base etc) */ +#define COMMAND_5C 0x5c /* */ +#define COMMAND_60 0x60 /* */ +#define COMMAND_66 0x66 /* */ +#define COMMAND_6C 0x6c /* */ +#define COMMAND_6E 0x6e /* */ +#define COMMAND_88 0x88 /* Unknown command */ +#define DSP_INIT_MSS 0x8c /* Enable Microsoft Sound System mode */ +#define COMMAND_C5 0xc5 /* */ +#define GET_DSP_VERSION 0xe1 /* Get DSP Version */ +#define GET_DSP_COPYRIGHT 0xe3 /* Get DSP Copyright */ + +/* + * Offsets of SC6000 DSP I/O ports. The offset is added to base I/O port + * to have the actual I/O port. + * Register permissions are: + * (wo) == Write Only + * (ro) == Read Only + * (w-) == Write + * (r-) == Read + */ +#define DSP_RESET 0x06 /* offset of DSP RESET (wo) */ +#define DSP_READ 0x0a /* offset of DSP READ (ro) */ +#define DSP_WRITE 0x0c /* offset of DSP WRITE (w-) */ +#define DSP_COMMAND 0x0c /* offset of DSP COMMAND (w-) */ +#define DSP_STATUS 0x0c /* offset of DSP STATUS (r-) */ +#define DSP_DATAVAIL 0x0e /* offset of DSP DATA AVAILABLE (ro) */ + +#define PFX "sc6000: " +#define DRV_NAME "SC-6000" + +/* hardware dependent functions */ + +/* + * sc6000_irq_to_softcfg - Decode irq number into cfg code. + */ +static __devinit unsigned char sc6000_irq_to_softcfg(int irq) +{ + unsigned char val = 0; + + switch (irq) { + case 5: + val = 0x28; + break; + case 7: + val = 0x8; + break; + case 9: + val = 0x10; + break; + case 10: + val = 0x18; + break; + case 11: + val = 0x20; + break; + default: + break; + } + return val; +} + +/* + * sc6000_dma_to_softcfg - Decode dma number into cfg code. + */ +static __devinit unsigned char sc6000_dma_to_softcfg(int dma) +{ + unsigned char val = 0; + + switch (dma) { + case 0: + val = 1; + break; + case 1: + val = 2; + break; + case 3: + val = 3; + break; + default: + break; + } + return val; +} + +/* + * sc6000_mpu_irq_to_softcfg - Decode MPU-401 irq number into cfg code. + */ +static __devinit unsigned char sc6000_mpu_irq_to_softcfg(int mpu_irq) +{ + unsigned char val = 0; + + switch (mpu_irq) { + case 5: + val = 4; + break; + case 7: + val = 0x44; + break; + case 9: + val = 0x84; + break; + case 10: + val = 0xc4; + break; + default: + break; + } + return val; +} + +static __devinit int sc6000_wait_data(char __iomem *vport) +{ + int loop = 1000; + unsigned char val = 0; + + do { + val = ioread8(vport + DSP_DATAVAIL); + if (val & 0x80) + return 0; + cpu_relax(); + } while (loop--); + + return -EAGAIN; +} + +static __devinit int sc6000_read(char __iomem *vport) +{ + if (sc6000_wait_data(vport)) + return -EBUSY; + + return ioread8(vport + DSP_READ); + +} + +static __devinit int sc6000_write(char __iomem *vport, int cmd) +{ + unsigned char val; + int loop = 500000; + + do { + val = ioread8(vport + DSP_STATUS); + /* + * DSP ready to receive data if bit 7 of val == 0 + */ + if (!(val & 0x80)) { + iowrite8(cmd, vport + DSP_COMMAND); + return 0; + } + cpu_relax(); + } while (loop--); + + snd_printk(KERN_ERR "DSP Command (0x%x) timeout.\n", cmd); + + return -EIO; +} + +static int __devinit sc6000_dsp_get_answer(char __iomem *vport, int command, + char *data, int data_len) +{ + int len = 0; + + if (sc6000_write(vport, command)) { + snd_printk(KERN_ERR "CMD 0x%x: failed!\n", command); + return -EIO; + } + + do { + int val = sc6000_read(vport); + + if (val < 0) + break; + + data[len++] = val; + + } while (len < data_len); + + /* + * If no more data available, return to the caller, no error if len>0. + * We have no other way to know when the string is finished. + */ + return len ? len : -EIO; +} + +static int __devinit sc6000_dsp_reset(char __iomem *vport) +{ + iowrite8(1, vport + DSP_RESET); + udelay(10); + iowrite8(0, vport + DSP_RESET); + udelay(20); + if (sc6000_read(vport) == 0xaa) + return 0; + return -ENODEV; +} + +/* detection and initialization */ +static int __devinit sc6000_cfg_write(char __iomem *vport, + unsigned char softcfg) +{ + + if (sc6000_write(vport, WRITE_MDIRQ_CFG)) { + snd_printk(KERN_ERR "CMD 0x%x: failed!\n", WRITE_MDIRQ_CFG); + return -EIO; + } + if (sc6000_write(vport, softcfg)) { + snd_printk(KERN_ERR "sc6000_cfg_write: failed!\n"); + return -EIO; + } + return 0; +} + +static int __devinit sc6000_setup_board(char __iomem *vport, int config) +{ + int loop = 10; + + do { + if (sc6000_write(vport, COMMAND_88)) { + snd_printk(KERN_ERR "CMD 0x%x: failed!\n", + COMMAND_88); + return -EIO; + } + } while ((sc6000_wait_data(vport) < 0) && loop--); + + if (sc6000_read(vport) < 0) { + snd_printk(KERN_ERR "sc6000_read after CMD 0x%x: failed\n", + COMMAND_88); + return -EIO; + } + + if (sc6000_cfg_write(vport, config)) + return -ENODEV; + + return 0; +} + +static int __devinit sc6000_init_mss(char __iomem *vport, int config, + char __iomem *vmss_port, int mss_config) +{ + if (sc6000_write(vport, DSP_INIT_MSS)) { + snd_printk(KERN_ERR "sc6000_init_mss [0x%x]: failed!\n", + DSP_INIT_MSS); + return -EIO; + } + + msleep(10); + + if (sc6000_cfg_write(vport, config)) + return -EIO; + + iowrite8(mss_config, vmss_port); + + return 0; +} + +static int __devinit sc6000_init_board(char __iomem *vport, int irq, int dma, + char __iomem *vmss_port, int mpu_irq) +{ + char answer[15]; + char version[2]; + int mss_config = sc6000_irq_to_softcfg(irq) | + sc6000_dma_to_softcfg(dma); + int config = mss_config | + sc6000_mpu_irq_to_softcfg(mpu_irq); + int err; + + err = sc6000_dsp_reset(vport); + if (err < 0) { + snd_printk(KERN_ERR "sc6000_dsp_reset: failed!\n"); + return err; + } + + memset(answer, 0, sizeof(answer)); + err = sc6000_dsp_get_answer(vport, GET_DSP_COPYRIGHT, answer, 15); + if (err <= 0) { + snd_printk(KERN_ERR "sc6000_dsp_copyright: failed!\n"); + return -ENODEV; + } + /* + * My SC-6000 card return "SC-6000" in DSPCopyright, so + * if we have something different, we have to be warned. + * Mine returns "SC-6000A " - KH + */ + if (strncmp("SC-6000", answer, 7)) + snd_printk(KERN_WARNING "Warning: non SC-6000 audio card!\n"); + + if (sc6000_dsp_get_answer(vport, GET_DSP_VERSION, version, 2) < 2) { + snd_printk(KERN_ERR "sc6000_dsp_version: failed!\n"); + return -ENODEV; + } + printk(KERN_INFO PFX "Detected model: %s, DSP version %d.%d\n", + answer, version[0], version[1]); + + /* + * 0x0A == (IRQ 7, DMA 1, MIRQ 0) + */ + err = sc6000_cfg_write(vport, 0x0a); + if (err < 0) { + snd_printk(KERN_ERR "sc6000_cfg_write: failed!\n"); + return -EFAULT; + } + + err = sc6000_setup_board(vport, config); + if (err < 0) { + snd_printk(KERN_ERR "sc6000_setup_board: failed!\n"); + return -ENODEV; + } + + err = sc6000_init_mss(vport, config, vmss_port, mss_config); + if (err < 0) { + snd_printk(KERN_ERR "Can not initialize" + "Microsoft Sound System mode.\n"); + return -ENODEV; + } + + return 0; +} + +static int __devinit snd_sc6000_mixer(struct snd_ad1848 *chip) +{ + struct snd_card *card = chip->card; + struct snd_ctl_elem_id id1, id2; + int err; + + memset(&id1, 0, sizeof(id1)); + memset(&id2, 0, sizeof(id2)); + id1.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + id2.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + /* reassign AUX0 to FM */ + strcpy(id1.name, "Aux Playback Switch"); + strcpy(id2.name, "FM Playback Switch"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) + return err; + strcpy(id1.name, "Aux Playback Volume"); + strcpy(id2.name, "FM Playback Volume"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) + return err; + /* reassign AUX1 to CD */ + strcpy(id1.name, "Aux Playback Switch"); id1.index = 1; + strcpy(id2.name, "CD Playback Switch"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) + return err; + strcpy(id1.name, "Aux Playback Volume"); + strcpy(id2.name, "CD Playback Volume"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) + return err; + return 0; +} + +static int __devinit snd_sc6000_match(struct device *devptr, unsigned int dev) +{ + if (!enable[dev]) + return 0; + if (port[dev] == SNDRV_AUTO_PORT) { + printk(KERN_ERR PFX "specify IO port\n"); + return 0; + } + if (mss_port[dev] == SNDRV_AUTO_PORT) { + printk(KERN_ERR PFX "specify MSS port\n"); + return 0; + } + if (port[dev] != 0x220 && port[dev] != 0x240) { + printk(KERN_ERR PFX "Port must be 0x220 or 0x240\n"); + return 0; + } + if (mss_port[dev] != 0x530 && mss_port[dev] != 0xe80) { + printk(KERN_ERR PFX "MSS port must be 0x530 or 0xe80\n"); + return 0; + } + if (irq[dev] != SNDRV_AUTO_IRQ && !sc6000_irq_to_softcfg(irq[dev])) { + printk(KERN_ERR PFX "invalid IRQ %d\n", irq[dev]); + return 0; + } + if (dma[dev] != SNDRV_AUTO_DMA && !sc6000_dma_to_softcfg(dma[dev])) { + printk(KERN_ERR PFX "invalid DMA %d\n", dma[dev]); + return 0; + } + if (mpu_port[dev] != SNDRV_AUTO_PORT && + (mpu_port[dev] & ~0x30L) != 0x300) { + printk(KERN_ERR PFX "invalid MPU-401 port %lx\n", + mpu_port[dev]); + return 0; + } + if (mpu_port[dev] != SNDRV_AUTO_PORT && + mpu_irq[dev] != SNDRV_AUTO_IRQ && mpu_irq[dev] != 0 && + !sc6000_mpu_irq_to_softcfg(mpu_irq[dev])) { + printk(KERN_ERR PFX "invalid MPU-401 IRQ %d\n", mpu_irq[dev]); + return 0; + } + return 1; +} + +static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev) +{ + static int possible_irqs[] = { 5, 7, 9, 10, 11, -1 }; + static int possible_dmas[] = { 1, 3, 0, -1 }; + int err; + int xirq = irq[dev]; + int xdma = dma[dev]; + struct snd_card *card; + struct snd_ad1848 *chip; + struct snd_opl3 *opl3; + char __iomem *vport; + char __iomem *vmss_port; + + + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + if (!card) + return -ENOMEM; + + if (xirq == SNDRV_AUTO_IRQ) { + xirq = snd_legacy_find_free_irq(possible_irqs); + if (xirq < 0) { + snd_printk(KERN_ERR PFX "unable to find a free IRQ\n"); + err = -EBUSY; + goto err_exit; + } + } + + if (xdma == SNDRV_AUTO_DMA) { + xdma = snd_legacy_find_free_dma(possible_dmas); + if (xdma < 0) { + snd_printk(KERN_ERR PFX "unable to find a free DMA\n"); + err = -EBUSY; + goto err_exit; + } + } + + if (!request_region(port[dev], 0x10, DRV_NAME)) { + snd_printk(KERN_ERR PFX + "I/O port region is already in use.\n"); + err = -EBUSY; + goto err_exit; + } + vport = devm_ioport_map(devptr, port[dev], 0x10); + if (!vport) { + snd_printk(KERN_ERR PFX + "I/O port cannot be iomaped.\n"); + err = -EBUSY; + goto err_unmap1; + } + + /* to make it marked as used */ + if (!request_region(mss_port[dev], 4, DRV_NAME)) { + snd_printk(KERN_ERR PFX + "SC-6000 port I/O port region is already in use.\n"); + err = -EBUSY; + goto err_unmap1; + } + vmss_port = devm_ioport_map(devptr, mss_port[dev], 4); + if (!vport) { + snd_printk(KERN_ERR PFX + "MSS port I/O cannot be iomaped.\n"); + err = -EBUSY; + goto err_unmap2; + } + + snd_printd("Initializing BASE[0x%lx] IRQ[%d] DMA[%d] MIRQ[%d]\n", + port[dev], xirq, xdma, + mpu_irq[dev] == SNDRV_AUTO_IRQ ? 0 : mpu_irq[dev]); + + err = sc6000_init_board(vport, xirq, xdma, vmss_port, mpu_irq[dev]); + if (err < 0) + goto err_unmap2; + + err = snd_ad1848_create(card, mss_port[dev] + 4, xirq, xdma, + AD1848_HW_DETECT, &chip); + if (err < 0) + goto err_unmap2; + card->private_data = chip; + + err = snd_ad1848_pcm(chip, 0, NULL); + if (err < 0) { + snd_printk(KERN_ERR PFX + "error creating new ad1848 PCM device\n"); + goto err_unmap2; + } + err = snd_ad1848_mixer(chip); + if (err < 0) { + snd_printk(KERN_ERR PFX "error creating new ad1848 mixer\n"); + goto err_unmap2; + } + err = snd_sc6000_mixer(chip); + if (err < 0) { + snd_printk(KERN_ERR PFX "the mixer rewrite failed\n"); + goto err_unmap2; + } + if (snd_opl3_create(card, + 0x388, 0x388 + 2, + OPL3_HW_AUTO, 0, &opl3) < 0) { + snd_printk(KERN_ERR PFX "no OPL device at 0x%x-0x%x ?\n", + 0x388, 0x388 + 2); + } else { + err = snd_opl3_timer_new(opl3, 0, 1); + if (err < 0) + goto err_unmap2; + + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) + goto err_unmap2; + } + + if (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_MPU401, + mpu_port[dev], 0, + mpu_irq[dev], IRQF_DISABLED, + NULL) < 0) + snd_printk(KERN_ERR "no MPU-401 device at 0x%lx ?\n", + mpu_port[dev]); + } + + strcpy(card->driver, DRV_NAME); + strcpy(card->shortname, "SC-6000"); + sprintf(card->longname, "Gallant SC-6000 at 0x%lx, irq %d, dma %d", + mss_port[dev], xirq, xdma); + + snd_card_set_dev(card, devptr); + + err = snd_card_register(card); + if (err < 0) + goto err_unmap2; + + dev_set_drvdata(devptr, card); + return 0; + +err_unmap2: + release_region(mss_port[dev], 4); +err_unmap1: + release_region(port[dev], 0x10); +err_exit: + snd_card_free(card); + return err; +} + +static int __devexit snd_sc6000_remove(struct device *devptr, unsigned int dev) +{ + release_region(port[dev], 0x10); + release_region(mss_port[dev], 4); + + snd_card_free(dev_get_drvdata(devptr)); + dev_set_drvdata(devptr, NULL); + return 0; +} + +static struct isa_driver snd_sc6000_driver = { + .match = snd_sc6000_match, + .probe = snd_sc6000_probe, + .remove = __devexit_p(snd_sc6000_remove), + /* FIXME: suspend/resume */ + .driver = { + .name = DRV_NAME, + }, +}; + + +static int __init alsa_card_sc6000_init(void) +{ + return isa_register_driver(&snd_sc6000_driver, SNDRV_CARDS); +} + +static void __exit alsa_card_sc6000_exit(void) +{ + isa_unregister_driver(&snd_sc6000_driver); +} + +module_init(alsa_card_sc6000_init) +module_exit(alsa_card_sc6000_exit) diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index cbad2a51cbaa..1cb921d6137e 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -45,10 +45,12 @@ MODULE_LICENSE("GPL"); static int index[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IDX; static char* id[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_STR; -static long port[SNDRV_CARDS] __devinitdata = { [0 ... (SNDRV_CARDS-1)] = SNDRV_AUTO_PORT }; +static long port[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT; +static long wss_port[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT; static int irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ; static int mpu_irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ; static int dma[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_DMA; +static int dma2[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_DMA; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index number for SoundScape soundcard"); @@ -59,6 +61,9 @@ MODULE_PARM_DESC(id, "Description for SoundScape card"); module_param_array(port, long, NULL, 0444); MODULE_PARM_DESC(port, "Port # for SoundScape driver."); +module_param_array(wss_port, long, NULL, 0444); +MODULE_PARM_DESC(wss_port, "WSS Port # for SoundScape driver."); + module_param_array(irq, int, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for SoundScape driver."); @@ -68,12 +73,16 @@ MODULE_PARM_DESC(mpu_irq, "MPU401 IRQ # for SoundScape driver."); module_param_array(dma, int, NULL, 0444); MODULE_PARM_DESC(dma, "DMA # for SoundScape driver."); +module_param_array(dma2, int, NULL, 0444); +MODULE_PARM_DESC(dma2, "DMA2 # for SoundScape driver."); + #ifdef CONFIG_PNP static int isa_registered; static int pnp_registered; static struct pnp_card_device_id sscape_pnpids[] = { - { .id = "ENS3081", .devs = { { "ENS0000" } } }, + { .id = "ENS3081", .devs = { { "ENS0000" } } }, /* Soundscape PnP */ + { .id = "ENS4081", .devs = { { "ENS1011" } } }, /* VIVO90 */ { .id = "" } /* end */ }; @@ -124,12 +133,21 @@ enum GA_REG { #define AD1845_FREQ_SEL_MSB 0x16 #define AD1845_FREQ_SEL_LSB 0x17 +enum card_type { + SSCAPE, + SSCAPE_PNP, + SSCAPE_VIVO, +}; + struct soundscape { spinlock_t lock; unsigned io_base; + unsigned wss_base; int codec_type; int ic_type; + enum card_type type; struct resource *io_res; + struct resource *wss_res; struct snd_cs4231 *chip; struct snd_mpu401 *mpu; struct snd_hwdep *hw; @@ -340,8 +358,9 @@ static inline void activate_ad1845_unsafe(unsigned io_base) */ static void soundscape_free(struct snd_card *c) { - register struct soundscape *sscape = get_card_soundscape(c); + struct soundscape *sscape = get_card_soundscape(c); release_and_free_resource(sscape->io_res); + release_and_free_resource(sscape->wss_res); free_dma(sscape->chip->dma1); } @@ -382,7 +401,7 @@ static int obp_startup_ack(struct soundscape *s, unsigned timeout) unsigned long flags; unsigned char x; - schedule_timeout(1); + schedule_timeout_uninterruptible(1); spin_lock_irqsave(&s->lock, flags); x = inb(HOST_DATA_IO(s->io_base)); @@ -409,7 +428,7 @@ static int host_startup_ack(struct soundscape *s, unsigned timeout) unsigned long flags; unsigned char x; - schedule_timeout(1); + schedule_timeout_uninterruptible(1); spin_lock_irqsave(&s->lock, flags); x = inb(HOST_DATA_IO(s->io_base)); @@ -522,7 +541,7 @@ static int upload_dma_data(struct soundscape *s, ret = -EAGAIN; } - _release_dma: +_release_dma: /* * NOTE!!! We are NOT holding any spinlocks at this point !!! */ @@ -802,6 +821,7 @@ static int __devinit detect_sscape(struct soundscape *s) unsigned long flags; unsigned d; int retval = 0; + int codec = s->wss_base; spin_lock_irqsave(&s->lock, flags); @@ -833,9 +853,27 @@ static int __devinit detect_sscape(struct soundscape *s) outb(0xfe, ODIE_ADDR_IO(s->io_base)); if ((inb(ODIE_ADDR_IO(s->io_base)) & 0x9f) != 0x0e) goto _done; - if ((inb(ODIE_DATA_IO(s->io_base)) & 0x9f) != 0x0e) + + outb(0xfe, ODIE_ADDR_IO(s->io_base)); + d = inb(ODIE_DATA_IO(s->io_base)); + if (s->type != SSCAPE_VIVO && (d & 0x9f) != 0x0e) goto _done; + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); + + if (s->type == SSCAPE_VIVO) + codec += 4; + /* wait for WSS codec */ + for (d = 0; d < 500; d++) { + if ((inb(codec) & 0x80) == 0) + break; + spin_unlock_irqrestore(&s->lock, flags); + msleep(1); + spin_lock_irqsave(&s->lock, flags); + } + snd_printd(KERN_INFO "init delay = %d ms\n", d); + /* * SoundScape successfully detected! */ @@ -995,21 +1033,23 @@ static void ad1845_capture_format(struct snd_cs4231 * chip, struct snd_pcm_hw_pa * try to support at least some of the extra bits by overriding * some of the CS4231 callback. */ -static int __devinit create_ad1845(struct snd_card *card, unsigned port, int irq, int dma1) +static int __devinit create_ad1845(struct snd_card *card, unsigned port, + int irq, int dma1, int dma2) { register struct soundscape *sscape = get_card_soundscape(card); struct snd_cs4231 *chip; int err; -#define CS4231_SHARE_HARDWARE (CS4231_HWSHARE_DMA1 | CS4231_HWSHARE_DMA2) - /* - * The AD1845 PCM device is only half-duplex, and so - * we only give it one DMA channel ... - */ - if ((err = snd_cs4231_create(card, - port, -1, irq, dma1, dma1, - CS4231_HW_DETECT, - CS4231_HWSHARE_DMA1, &chip)) == 0) { + if (sscape->type == SSCAPE_VIVO) + port += 4; + + if (dma1 == dma2) + dma2 = -1; + + err = snd_cs4231_create(card, + port, -1, irq, dma1, dma2, + CS4231_HW_DETECT, CS4231_HWSHARE_DMA1, &chip); + if (!err) { unsigned long flags; struct snd_pcm *pcm; @@ -1031,49 +1071,72 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, int irq snd_cs4231_mce_down(chip); */ - /* - * The input clock frequency on the SoundScape must - * be 14.31818 MHz, because we must set this register - * to get the playback to sound correct ... - */ - snd_cs4231_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - snd_cs4231_out(chip, AD1845_CRYS_CLOCK_SEL, 0x20); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_cs4231_mce_down(chip); + if (sscape->type != SSCAPE_VIVO) { + int val; + /* + * The input clock frequency on the SoundScape must + * be 14.31818 MHz, because we must set this register + * to get the playback to sound correct ... + */ + snd_cs4231_mce_up(chip); + spin_lock_irqsave(&chip->reg_lock, flags); + snd_cs4231_out(chip, AD1845_CRYS_CLOCK_SEL, 0x20); + spin_unlock_irqrestore(&chip->reg_lock, flags); + snd_cs4231_mce_down(chip); - /* - * More custom configuration: - * a) select "mode 2", and provide a current drive of 8 mA - * b) enable frequency selection (for capture/playback) - */ - spin_lock_irqsave(&chip->reg_lock, flags); - snd_cs4231_out(chip, CS4231_MISC_INFO, (CS4231_MODE2 | 0x10)); - snd_cs4231_out(chip, AD1845_PWR_DOWN_CTRL, snd_cs4231_in(chip, AD1845_PWR_DOWN_CTRL) | AD1845_FREQ_SEL_ENABLE); - spin_unlock_irqrestore(&chip->reg_lock, flags); + /* + * More custom configuration: + * a) select "mode 2" and provide a current drive of 8mA + * b) enable frequency selection (for capture/playback) + */ + spin_lock_irqsave(&chip->reg_lock, flags); + snd_cs4231_out(chip, CS4231_MISC_INFO, + CS4231_MODE2 | 0x10); + val = snd_cs4231_in(chip, AD1845_PWR_DOWN_CTRL); + snd_cs4231_out(chip, AD1845_PWR_DOWN_CTRL, + val | AD1845_FREQ_SEL_ENABLE); + spin_unlock_irqrestore(&chip->reg_lock, flags); + } - if ((err = snd_cs4231_pcm(chip, 0, &pcm)) < 0) { - snd_printk(KERN_ERR "sscape: No PCM device for AD1845 chip\n"); + err = snd_cs4231_pcm(chip, 0, &pcm); + if (err < 0) { + snd_printk(KERN_ERR "sscape: No PCM device " + "for AD1845 chip\n"); goto _error; } - if ((err = snd_cs4231_mixer(chip)) < 0) { - snd_printk(KERN_ERR "sscape: No mixer device for AD1845 chip\n"); + err = snd_cs4231_mixer(chip); + if (err < 0) { + snd_printk(KERN_ERR "sscape: No mixer device " + "for AD1845 chip\n"); goto _error; } - - if ((err = snd_ctl_add(card, snd_ctl_new1(&midi_mixer_ctl, chip))) < 0) { - snd_printk(KERN_ERR "sscape: Could not create MIDI mixer control\n"); + err = snd_cs4231_timer(chip, 0, NULL); + if (err < 0) { + snd_printk(KERN_ERR "sscape: No timer device " + "for AD1845 chip\n"); goto _error; } + if (sscape->type != SSCAPE_VIVO) { + err = snd_ctl_add(card, + snd_ctl_new1(&midi_mixer_ctl, chip)); + if (err < 0) { + snd_printk(KERN_ERR "sscape: Could not create " + "MIDI mixer control\n"); + goto _error; + } + chip->set_playback_format = ad1845_playback_format; + chip->set_capture_format = ad1845_capture_format; + } + strcpy(card->driver, "SoundScape"); strcpy(card->shortname, pcm->name); snprintf(card->longname, sizeof(card->longname), - "%s at 0x%lx, IRQ %d, DMA %d\n", - pcm->name, chip->port, chip->irq, chip->dma1); - chip->set_playback_format = ad1845_playback_format; - chip->set_capture_format = ad1845_capture_format; + "%s at 0x%lx, IRQ %d, DMA1 %d, DMA2 %d\n", + pcm->name, chip->port, chip->irq, + chip->dma1, chip->dma2); + sscape->chip = chip; } @@ -1086,15 +1149,15 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, int irq * Create an ALSA soundcard entry for the SoundScape, using * the given list of port, IRQ and DMA resources. */ -static int __devinit create_sscape(int dev, struct snd_card **rcardp) +static int __devinit create_sscape(int dev, struct snd_card *card) { - struct snd_card *card; - register struct soundscape *sscape; - register unsigned dma_cfg; + struct soundscape *sscape = get_card_soundscape(card); + unsigned dma_cfg; unsigned irq_cfg; unsigned mpu_irq_cfg; unsigned xport; struct resource *io_res; + struct resource *wss_res; unsigned long flags; int err; @@ -1118,61 +1181,69 @@ static int __devinit create_sscape(int dev, struct snd_card **rcardp) * Grab IO ports that we will need to probe so that we * can detect and control this hardware ... */ - if ((io_res = request_region(xport, 8, "SoundScape")) == NULL) { + io_res = request_region(xport, 8, "SoundScape"); + if (!io_res) { snd_printk(KERN_ERR "sscape: can't grab port 0x%x\n", xport); return -EBUSY; } + wss_res = NULL; + if (sscape->type == SSCAPE_VIVO) { + wss_res = request_region(wss_port[dev], 4, "SoundScape"); + if (!wss_res) { + snd_printk(KERN_ERR "sscape: can't grab port 0x%lx\n", + wss_port[dev]); + err = -EBUSY; + goto _release_region; + } + } /* - * Grab both DMA channels (OK, only one for now) ... + * Grab one DMA channel ... */ - if ((err = request_dma(dma[dev], "SoundScape")) < 0) { + err = request_dma(dma[dev], "SoundScape"); + if (err < 0) { snd_printk(KERN_ERR "sscape: can't grab DMA %d\n", dma[dev]); goto _release_region; } - /* - * Create a new ALSA sound card entry, in anticipation - * of detecting our hardware ... - */ - if ((card = snd_card_new(index[dev], id[dev], THIS_MODULE, - sizeof(struct soundscape))) == NULL) { - err = -ENOMEM; - goto _release_dma; - } - - sscape = get_card_soundscape(card); spin_lock_init(&sscape->lock); spin_lock_init(&sscape->fwlock); sscape->io_res = io_res; + sscape->wss_res = wss_res; sscape->io_base = xport; + sscape->wss_base = wss_port[dev]; if (!detect_sscape(sscape)) { printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", sscape->io_base); err = -ENODEV; - goto _release_card; + goto _release_dma; } printk(KERN_INFO "sscape: hardware detected at 0x%x, using IRQ %d, DMA %d\n", - sscape->io_base, irq[dev], dma[dev]); + sscape->io_base, irq[dev], dma[dev]); - /* - * Now create the hardware-specific device so that we can - * load the microcode into the on-board processor. - * We cannot use the MPU-401 MIDI system until this firmware - * has been loaded into the card. - */ - if ((err = snd_hwdep_new(card, "MC68EC000", 0, &(sscape->hw))) < 0) { - printk(KERN_ERR "sscape: Failed to create firmware device\n"); - goto _release_card; + if (sscape->type != SSCAPE_VIVO) { + /* + * Now create the hardware-specific device so that we can + * load the microcode into the on-board processor. + * We cannot use the MPU-401 MIDI system until this firmware + * has been loaded into the card. + */ + err = snd_hwdep_new(card, "MC68EC000", 0, &(sscape->hw)); + if (err < 0) { + printk(KERN_ERR "sscape: Failed to create " + "firmware device\n"); + goto _release_dma; + } + strlcpy(sscape->hw->name, "SoundScape M68K", + sizeof(sscape->hw->name)); + sscape->hw->name[sizeof(sscape->hw->name) - 1] = '\0'; + sscape->hw->iface = SNDRV_HWDEP_IFACE_SSCAPE; + sscape->hw->ops.open = sscape_hw_open; + sscape->hw->ops.release = sscape_hw_release; + sscape->hw->ops.ioctl = sscape_hw_ioctl; + sscape->hw->private_data = sscape; } - strlcpy(sscape->hw->name, "SoundScape M68K", sizeof(sscape->hw->name)); - sscape->hw->name[sizeof(sscape->hw->name) - 1] = '\0'; - sscape->hw->iface = SNDRV_HWDEP_IFACE_SSCAPE; - sscape->hw->ops.open = sscape_hw_open; - sscape->hw->ops.release = sscape_hw_release; - sscape->hw->ops.ioctl = sscape_hw_ioctl; - sscape->hw->private_data = sscape; /* * Tell the on-board devices where their resources are (I think - @@ -1197,7 +1268,8 @@ static int __devinit create_sscape(int dev, struct snd_card **rcardp) sscape_write_unsafe(sscape->io_base, GA_INTCFG_REG, 0xf0 | (mpu_irq_cfg << 2) | mpu_irq_cfg); sscape_write_unsafe(sscape->io_base, - GA_CDCFG_REG, 0x09 | DMA_8BIT | (dma[dev] << 4) | (irq_cfg << 1)); + GA_CDCFG_REG, 0x09 | DMA_8BIT + | (dma[dev] << 4) | (irq_cfg << 1)); spin_unlock_irqrestore(&sscape->lock, flags); @@ -1205,30 +1277,37 @@ static int __devinit create_sscape(int dev, struct snd_card **rcardp) * We have now enabled the codec chip, and so we should * detect the AD1845 device ... */ - if ((err = create_ad1845(card, CODEC_IO(xport), irq[dev], dma[dev])) < 0) { - printk(KERN_ERR "sscape: No AD1845 device at 0x%x, IRQ %d\n", - CODEC_IO(xport), irq[dev]); - goto _release_card; + err = create_ad1845(card, wss_port[dev], irq[dev], + dma[dev], dma2[dev]); + if (err < 0) { + printk(KERN_ERR "sscape: No AD1845 device at 0x%lx, IRQ %d\n", + wss_port[dev], irq[dev]); + goto _release_dma; } #define MIDI_DEVNUM 0 - if ((err = create_mpu401(card, MIDI_DEVNUM, MPU401_IO(xport), mpu_irq[dev])) < 0) { - printk(KERN_ERR "sscape: Failed to create MPU-401 device at 0x%x\n", - MPU401_IO(xport)); - goto _release_card; - } + if (sscape->type != SSCAPE_VIVO) { + err = create_mpu401(card, MIDI_DEVNUM, + MPU401_IO(xport), mpu_irq[dev]); + if (err < 0) { + printk(KERN_ERR "sscape: Failed to create " + "MPU-401 device at 0x%x\n", + MPU401_IO(xport)); + goto _release_dma; + } - /* - * Enable the master IRQ ... - */ - sscape_write(sscape, GA_INTENA_REG, 0x80); + /* + * Enable the master IRQ ... + */ + sscape_write(sscape, GA_INTENA_REG, 0x80); - /* - * Initialize mixer - */ - sscape->midi_vol = 0; - host_write_ctrl_unsafe(sscape->io_base, CMD_SET_MIDI_VOL, 100); - host_write_ctrl_unsafe(sscape->io_base, 0, 100); - host_write_ctrl_unsafe(sscape->io_base, CMD_XXX_MIDI_VOL, 100); + /* + * Initialize mixer + */ + sscape->midi_vol = 0; + host_write_ctrl_unsafe(sscape->io_base, CMD_SET_MIDI_VOL, 100); + host_write_ctrl_unsafe(sscape->io_base, 0, 100); + host_write_ctrl_unsafe(sscape->io_base, CMD_XXX_MIDI_VOL, 100); + } /* * Now that we have successfully created this sound card, @@ -1237,17 +1316,14 @@ static int __devinit create_sscape(int dev, struct snd_card **rcardp) * function now that our "constructor" has completed. */ card->private_free = soundscape_free; - *rcardp = card; return 0; - _release_card: - snd_card_free(card); - - _release_dma: +_release_dma: free_dma(dma[dev]); - _release_region: +_release_region: + release_and_free_resource(wss_res); release_and_free_resource(io_res); return err; @@ -1276,19 +1352,33 @@ static int __devinit snd_sscape_match(struct device *pdev, unsigned int i) static int __devinit snd_sscape_probe(struct device *pdev, unsigned int dev) { struct snd_card *card; + struct soundscape *sscape; int ret; + card = snd_card_new(index[dev], id[dev], THIS_MODULE, + sizeof(struct soundscape)); + if (!card) + return -ENOMEM; + + sscape = get_card_soundscape(card); + sscape->type = SSCAPE; + dma[dev] &= 0x03; - ret = create_sscape(dev, &card); + ret = create_sscape(dev, card); if (ret < 0) - return ret; + goto _release_card; + snd_card_set_dev(card, pdev); if ((ret = snd_card_register(card)) < 0) { printk(KERN_ERR "sscape: Failed to register sound card\n"); - return ret; + goto _release_card; } dev_set_drvdata(pdev, card); return 0; + +_release_card: + snd_card_free(card); + return ret; } static int __devexit snd_sscape_remove(struct device *devptr, unsigned int dev) @@ -1325,6 +1415,7 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, static int idx = 0; struct pnp_dev *dev; struct snd_card *card; + struct soundscape *sscape; int ret; /* @@ -1366,26 +1457,55 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, } /* + * Create a new ALSA sound card entry, in anticipation + * of detecting our hardware ... + */ + card = snd_card_new(index[idx], id[idx], THIS_MODULE, + sizeof(struct soundscape)); + if (!card) + return -ENOMEM; + + sscape = get_card_soundscape(card); + + /* + * Identify card model ... + */ + if (!strncmp("ENS4081", pid->id, 7)) + sscape->type = SSCAPE_VIVO; + else + sscape->type = SSCAPE_PNP; + + /* * Read the correct parameters off the ISA PnP bus ... */ port[idx] = pnp_port_start(dev, 0); irq[idx] = pnp_irq(dev, 0); mpu_irq[idx] = pnp_irq(dev, 1); dma[idx] = pnp_dma(dev, 0) & 0x03; + if (sscape->type == SSCAPE_PNP) { + dma2[idx] = dma[idx]; + wss_port[idx] = CODEC_IO(port[idx]); + } else { + wss_port[idx] = pnp_port_start(dev, 1); + dma2[idx] = pnp_dma(dev, 1); + } - ret = create_sscape(idx, &card); + ret = create_sscape(idx, card); if (ret < 0) - return ret; + goto _release_card; + snd_card_set_dev(card, &pcard->card->dev); if ((ret = snd_card_register(card)) < 0) { printk(KERN_ERR "sscape: Failed to register sound card\n"); - snd_card_free(card); - return ret; + goto _release_card; } pnp_set_card_drvdata(pcard, card); ++idx; + return 0; +_release_card: + snd_card_free(card); return ret; } diff --git a/sound/isa/wavefront/Makefile b/sound/isa/wavefront/Makefile index b4cb28422db0..601bdddd44d0 100644 --- a/sound/isa/wavefront/Makefile +++ b/sound/isa/wavefront/Makefile @@ -1,6 +1,6 @@ # # Makefile for ALSA -# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz> +# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz> # snd-wavefront-objs := wavefront.o wavefront_fx.o wavefront_synth.o wavefront_midi.o diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index bacc51c86587..a1ebb7c5c684 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -27,6 +27,7 @@ #include <linux/delay.h> #include <linux/time.h> #include <linux/wait.h> +#include <linux/firmware.h> #include <linux/moduleparam.h> #include <sound/core.h> #include <sound/snd_wavefront.h> @@ -53,9 +54,8 @@ static int debug_default = 0; /* you can set this to control debugging /* XXX this needs to be made firmware and hardware version dependent */ -static char *ospath = "/etc/sound/wavefront.os"; /* where to find a processed - version of the WaveFront OS - */ +#define DEFAULT_OSPATH "wavefront.os" +static char *ospath = DEFAULT_OSPATH; /* the firmware file name */ static int wait_usecs = 150; /* This magic number seems to give pretty optimal throughput based on my limited experimentation. @@ -97,7 +97,7 @@ MODULE_PARM_DESC(sleep_interval, "how long to sleep when waiting for reply"); module_param(sleep_tries, int, 0444); MODULE_PARM_DESC(sleep_tries, "how many times to try sleeping during a wait"); module_param(ospath, charp, 0444); -MODULE_PARM_DESC(ospath, "full pathname to processed ICS2115 OS firmware"); +MODULE_PARM_DESC(ospath, "pathname to processed ICS2115 OS firmware"); module_param(reset_time, int, 0444); MODULE_PARM_DESC(reset_time, "how long to wait for a reset to take effect"); module_param(ramcheck_time, int, 0444); @@ -1768,7 +1768,7 @@ snd_wavefront_interrupt_bits (int irq) static void __devinit wavefront_should_cause_interrupt (snd_wavefront_t *dev, - int val, int port, int timeout) + int val, int port, unsigned long timeout) { wait_queue_t wait; @@ -1779,11 +1779,9 @@ wavefront_should_cause_interrupt (snd_wavefront_t *dev, dev->irq_ok = 0; outb (val,port); spin_unlock_irq(&dev->irq_lock); - while (1) { - if ((timeout = schedule_timeout(timeout)) == 0) - return; - if (dev->irq_ok) - return; + while (!dev->irq_ok && time_before(jiffies, timeout)) { + schedule_timeout_uninterruptible(1); + barrier(); } } @@ -1938,111 +1936,75 @@ wavefront_reset_to_cleanliness (snd_wavefront_t *dev) return (1); } -#include <linux/fs.h> -#include <linux/mm.h> -#include <linux/slab.h> -#include <linux/unistd.h> -#include <linux/syscalls.h> -#include <asm/uaccess.h> - - static int __devinit wavefront_download_firmware (snd_wavefront_t *dev, char *path) { - unsigned char section[WF_SECTION_MAX]; - signed char section_length; /* yes, just a char; max value is WF_SECTION_MAX */ + unsigned char *buf; + int len, err; int section_cnt_downloaded = 0; - int fd; - int c; - int i; - mm_segment_t fs; - - /* This tries to be a bit cleverer than the stuff Alan Cox did for - the generic sound firmware, in that it actually knows - something about the structure of the Motorola firmware. In - particular, it uses a version that has been stripped of the - 20K of useless header information, and had section lengths - added, making it possible to load the entire OS without any - [kv]malloc() activity, since the longest entity we ever read is - 42 bytes (well, WF_SECTION_MAX) long. - */ - - fs = get_fs(); - set_fs (get_ds()); + const struct firmware *firmware; - if ((fd = sys_open ((char __user *) path, 0, 0)) < 0) { - snd_printk ("Unable to load \"%s\".\n", - path); + err = request_firmware(&firmware, path, dev->card->dev); + if (err < 0) { + snd_printk(KERN_ERR "firmware (%s) download failed!!!\n", path); return 1; } - while (1) { - int x; - - if ((x = sys_read (fd, (char __user *) §ion_length, sizeof (section_length))) != - sizeof (section_length)) { - snd_printk ("firmware read error.\n"); - goto failure; - } - - if (section_length == 0) { + len = 0; + buf = firmware->data; + for (;;) { + int section_length = *(signed char *)buf; + if (section_length == 0) break; - } - if (section_length < 0 || section_length > WF_SECTION_MAX) { - snd_printk ("invalid firmware section length %d\n", - section_length); + snd_printk(KERN_ERR + "invalid firmware section length %d\n", + section_length); goto failure; } + buf++; + len++; - if (sys_read (fd, (char __user *) section, section_length) != section_length) { - snd_printk ("firmware section " - "read error.\n"); + if (firmware->size < len + section_length) { + snd_printk(KERN_ERR "firmware section read error.\n"); goto failure; } /* Send command */ - - if (wavefront_write (dev, WFC_DOWNLOAD_OS)) { + if (wavefront_write(dev, WFC_DOWNLOAD_OS)) goto failure; - } - for (i = 0; i < section_length; i++) { - if (wavefront_write (dev, section[i])) { + for (; section_length; section_length--) { + if (wavefront_write(dev, *buf)) goto failure; - } + buf++; + len++; } /* get ACK */ - - if (wavefront_wait (dev, STAT_CAN_READ)) { - - if ((c = inb (dev->data_port)) != WF_ACK) { - - snd_printk ("download " - "of section #%d not " - "acknowledged, ack = 0x%x\n", - section_cnt_downloaded + 1, c); - goto failure; - - } - - } else { - snd_printk ("time out for firmware ACK.\n"); + if (!wavefront_wait(dev, STAT_CAN_READ)) { + snd_printk(KERN_ERR "time out for firmware ACK.\n"); + goto failure; + } + err = inb(dev->data_port); + if (err != WF_ACK) { + snd_printk(KERN_ERR + "download of section #%d not " + "acknowledged, ack = 0x%x\n", + section_cnt_downloaded + 1, err); goto failure; } + section_cnt_downloaded++; } - sys_close (fd); - set_fs (fs); + release_firmware(firmware); return 0; failure: - sys_close (fd); - set_fs (fs); - snd_printk ("firmware download failed!!!\n"); + release_firmware(firmware); + snd_printk(KERN_ERR "firmware download failed!!!\n"); return 1; } @@ -2232,3 +2194,5 @@ snd_wavefront_detect (snd_wavefront_card_t *card) return 0; } + +MODULE_FIRMWARE(DEFAULT_OSPATH); diff --git a/sound/last.c b/sound/last.c index 964314efff5c..282b0cdb0589 100644 --- a/sound/last.c +++ b/sound/last.c @@ -1,6 +1,6 @@ /* * Advanced Linux Sound Architecture - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index c6b44102aa5b..356bf21a1506 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -170,14 +170,14 @@ config SND_CA0106 will be called snd-ca0106. config SND_CMIPCI - tristate "C-Media 8738, 8338" + tristate "C-Media 8338, 8738, 8768, 8770" depends on SND select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM help - If you want to use soundcards based on C-Media CMI8338 or CMI8738 - chips, say Y here and read + If you want to use soundcards based on C-Media CMI8338, CMI8738, + CMI8768 or CMI8770 chips, say Y here and read <file:Documentation/sound/alsa/CMIPCI.txt>. To compile this driver as a module, choose M here: the module @@ -500,6 +500,103 @@ config SND_HDA_INTEL To compile this driver as a module, choose M here: the module will be called snd-hda-intel. +config SND_HDA_HWDEP + bool "Build hwdep interface for HD-audio driver" + depends on SND_HDA_INTEL + select SND_HWDEP + help + Say Y here to build a hwdep interface for HD-audio driver. + This interface can be used for out-of-band communication + with codecs for debugging purposes. + +config SND_HDA_CODEC_REALTEK + bool "Build Realtek HD-audio codec support" + depends on SND_HDA_INTEL + default y + help + Say Y here to include Realtek HD-audio codec support in + snd-hda-intel driver, such as ALC880. + +config SND_HDA_CODEC_ANALOG + bool "Build Analog Device HD-audio codec support" + depends on SND_HDA_INTEL + default y + help + Say Y here to include Analog Device HD-audio codec support in + snd-hda-intel driver, such as AD1986A. + +config SND_HDA_CODEC_SIGMATEL + bool "Build IDT/Sigmatel HD-audio codec support" + depends on SND_HDA_INTEL + default y + help + Say Y here to include IDT (Sigmatel) HD-audio codec support in + snd-hda-intel driver, such as STAC9200. + +config SND_HDA_CODEC_VIA + bool "Build VIA HD-audio codec support" + depends on SND_HDA_INTEL + default y + help + Say Y here to include VIA HD-audio codec support in + snd-hda-intel driver, such as VT1708. + +config SND_HDA_CODEC_ATIHDMI + bool "Build ATI HDMI HD-audio codec support" + depends on SND_HDA_INTEL + default y + help + Say Y here to include ATI HDMI HD-audio codec support in + snd-hda-intel driver, such as ATI RS600 HDMI. + +config SND_HDA_CODEC_CONEXANT + bool "Build Conexant HD-audio codec support" + depends on SND_HDA_INTEL + default y + help + Say Y here to include Conexant HD-audio codec support in + snd-hda-intel driver, such as CX20549. + +config SND_HDA_CODEC_CMEDIA + bool "Build C-Media HD-audio codec support" + depends on SND_HDA_INTEL + default y + help + Say Y here to include C-Media HD-audio codec support in + snd-hda-intel driver, such as CMI9880. + +config SND_HDA_CODEC_SI3054 + bool "Build Silicon Labs 3054 HD-modem codec support" + depends on SND_HDA_INTEL + default y + help + Say Y here to include Silicon Labs 3054 HD-modem codec + (and compatibles) support in snd-hda-intel driver. + +config SND_HDA_GENERIC + bool "Enable generic HD-audio codec parser" + depends on SND_HDA_INTEL + default y + help + Say Y here to enable the generic HD-audio codec parser + in snd-hda-intel driver. + +config SND_HDA_POWER_SAVE + bool "Aggressive power-saving on HD-audio" + depends on SND_HDA_INTEL && EXPERIMENTAL + help + Say Y here to enable more aggressive power-saving mode on + HD-audio driver. The power-saving timeout can be configured + via power_save option or over sysfs on-the-fly. + +config SND_HDA_POWER_SAVE_DEFAULT + int "Default time-out for HD-audio power-save mode" + depends on SND_HDA_POWER_SAVE + default 0 + help + The default time-out value in seconds for HD-audio automatic + power-save mode. 0 means to disable the power-save mode. + config SND_HDSP tristate "RME Hammerfall DSP Audio" depends on SND @@ -799,4 +896,12 @@ config SND_AC97_POWER_SAVE snd-ac97-codec driver. You can toggle it dynamically over sysfs, too. +config SND_AC97_POWER_SAVE_DEFAULT + int "Default time-out for AC97 power-save mode" + depends on SND_AC97_POWER_SAVE + default 0 + help + The default time-out value in seconds for AC97 automatic + power-save mode. 0 means to disable the power-save mode. + endmenu diff --git a/sound/pci/Makefile b/sound/pci/Makefile index cd76e0293d06..09ddc82eeca2 100644 --- a/sound/pci/Makefile +++ b/sound/pci/Makefile @@ -1,6 +1,6 @@ # # Makefile for ALSA -# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz> +# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz> # snd-ad1889-objs := ad1889.o diff --git a/sound/pci/ac97/Makefile b/sound/pci/ac97/Makefile index f5d471896b95..0be48b1a22d0 100644 --- a/sound/pci/ac97/Makefile +++ b/sound/pci/ac97/Makefile @@ -1,6 +1,6 @@ # # Makefile for ALSA -# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz> +# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz> # snd-ac97-codec-objs := ac97_codec.o ac97_pcm.o diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index bbed644bf9c5..6a9966df0cc9 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Universal interface for Audio Codec '97 * * For more details look to AC '97 component specification revision 2.2 @@ -39,7 +39,7 @@ #include "ac97_patch.c" -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Universal interface for Audio Codec '97"); MODULE_LICENSE("GPL"); @@ -49,7 +49,7 @@ module_param(enable_loopback, bool, 0444); MODULE_PARM_DESC(enable_loopback, "Enable AC97 ADC/DAC Loopback Control"); #ifdef CONFIG_SND_AC97_POWER_SAVE -static int power_save; +static int power_save = CONFIG_SND_AC97_POWER_SAVE_DEFAULT; module_param(power_save, bool, 0644); MODULE_PARM_DESC(power_save, "Enable AC97 power-saving control"); #endif @@ -176,7 +176,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x574d4C09, 0xffffffff, "WM9709", NULL, NULL}, { 0x574d4C12, 0xffffffff, "WM9711,WM9712", patch_wolfson11, NULL}, { 0x574d4c13, 0xffffffff, "WM9713,WM9714", patch_wolfson13, NULL, AC97_DEFAULT_POWER_OFF}, -{ 0x594d4800, 0xffffffff, "YMF743", NULL, NULL }, +{ 0x594d4800, 0xffffffff, "YMF743", patch_yamaha_ymf743, NULL }, { 0x594d4802, 0xffffffff, "YMF752", NULL, NULL }, { 0x594d4803, 0xffffffff, "YMF753", patch_yamaha_ymf753, NULL }, { 0x83847600, 0xffffffff, "STAC9700,83,84", patch_sigmatel_stac9700, NULL }, @@ -779,6 +779,12 @@ static int snd_ac97_spdif_default_put(struct snd_kcontrol *kcontrol, struct snd_ change |= snd_ac97_update_bits_nolock(ac97, AC97_CXR_AUDIO_MISC, AC97_CXR_SPDIF_MASK | AC97_CXR_COPYRGT, v); + } else if (ac97->id == AC97_ID_YMF743) { + change |= snd_ac97_update_bits_nolock(ac97, + AC97_YMF7X3_DIT_CTRL, + 0xff38, + ((val << 4) & 0xff00) | + ((val << 2) & 0x0038)); } else { unsigned short extst = snd_ac97_read_cache(ac97, AC97_EXTENDED_STATUS); snd_ac97_update_bits_nolock(ac97, AC97_EXTENDED_STATUS, AC97_EA_SPDIF, 0); /* turn off */ @@ -1375,7 +1381,8 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) for (idx = 0; idx < 2; idx++) { if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_tone[idx], ac97))) < 0) return err; - if (ac97->id == AC97_ID_YMF753) { + if (ac97->id == AC97_ID_YMF743 || + ac97->id == AC97_ID_YMF753) { kctl->private_value &= ~(0xff << 16); kctl->private_value |= 7 << 16; } @@ -2036,11 +2043,12 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, else { udelay(50); if (ac97->scaps & AC97_SCAP_SKIP_AUDIO) - err = ac97_reset_wait(ac97, HZ/2, 1); + err = ac97_reset_wait(ac97, msecs_to_jiffies(500), 1); else { - err = ac97_reset_wait(ac97, HZ/2, 0); + err = ac97_reset_wait(ac97, msecs_to_jiffies(500), 0); if (err < 0) - err = ac97_reset_wait(ac97, HZ/2, 1); + err = ac97_reset_wait(ac97, + msecs_to_jiffies(500), 1); } if (err < 0) { snd_printk(KERN_WARNING "AC'97 %d does not respond - RESET\n", ac97->num); @@ -2104,7 +2112,7 @@ 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 + (HZ / 10); + end_time = jiffies + msecs_to_jiffies(100); do { if ((snd_ac97_read(ac97, AC97_POWERDOWN) & 0x0f) == 0x0f) goto __ready_ok; @@ -2136,7 +2144,7 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, udelay(100); /* nothing should be in powerdown mode */ snd_ac97_write_cache(ac97, AC97_EXTENDED_MSTATUS, 0); - end_time = jiffies + (HZ / 10); + end_time = jiffies + msecs_to_jiffies(100); do { if ((snd_ac97_read(ac97, AC97_EXTENDED_MSTATUS) & tmp) == tmp) goto __ready_ok; @@ -2354,7 +2362,8 @@ int snd_ac97_update_power(struct snd_ac97 *ac97, int reg, int powerup) * (for avoiding loud click noises for many (OSS) apps * that open/close frequently) */ - schedule_delayed_work(&ac97->power_work, HZ*2); + schedule_delayed_work(&ac97->power_work, + msecs_to_jiffies(2000)); else { cancel_delayed_work(&ac97->power_work); update_power_regs(ac97); @@ -2436,7 +2445,7 @@ EXPORT_SYMBOL(snd_ac97_suspend); /* * restore ac97 status */ -void snd_ac97_restore_status(struct snd_ac97 *ac97) +static void snd_ac97_restore_status(struct snd_ac97 *ac97) { int i; @@ -2457,7 +2466,7 @@ void snd_ac97_restore_status(struct snd_ac97 *ac97) /* * restore IEC958 status */ -void snd_ac97_restore_iec958(struct snd_ac97 *ac97) +static void snd_ac97_restore_iec958(struct snd_ac97 *ac97) { if (ac97->ext_id & AC97_EI_SPDIF) { if (ac97->regs[AC97_EXTENDED_STATUS] & AC97_EA_SPDIF) { @@ -2494,7 +2503,10 @@ void snd_ac97_resume(struct snd_ac97 *ac97) snd_ac97_write(ac97, AC97_POWERDOWN, 0); if (! (ac97->flags & AC97_DEFAULT_POWER_OFF)) { - snd_ac97_write(ac97, AC97_RESET, 0); + if (!(ac97->scaps & AC97_SCAP_SKIP_AUDIO)) + snd_ac97_write(ac97, AC97_RESET, 0); + else if (!(ac97->scaps & AC97_SCAP_SKIP_MODEM)) + snd_ac97_write(ac97, AC97_EXTENDED_MID, 0); udelay(100); snd_ac97_write(ac97, AC97_POWERDOWN, 0); } diff --git a/sound/pci/ac97/ac97_id.h b/sound/pci/ac97/ac97_id.h index 6d73514dc49e..c129492c82b3 100644 --- a/sound/pci/ac97/ac97_id.h +++ b/sound/pci/ac97/ac97_id.h @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Universal interface for Audio Codec '97 * * For more details look to AC '97 component specification revision 2.2 @@ -54,6 +54,7 @@ #define AC97_ID_ALC658 0x414c4780 #define AC97_ID_ALC658D 0x414c4781 #define AC97_ID_ALC850 0x414c4790 +#define AC97_ID_YMF743 0x594d4800 #define AC97_ID_YMF753 0x594d4803 #define AC97_ID_VT1616 0x49434551 #define AC97_ID_CM9738 0x434d4941 diff --git a/sound/pci/ac97/ac97_local.h b/sound/pci/ac97/ac97_local.h index 78745c5c6df8..c276a5e3f7ac 100644 --- a/sound/pci/ac97/ac97_local.h +++ b/sound/pci/ac97/ac97_local.h @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Universal interface for Audio Codec '97 * * For more details look to AC '97 component specification revision 2.2 diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 581ebba4d1a7..98c8b727b62b 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Universal interface for Audio Codec '97 * * For more details look to AC '97 component specification revision 2.2 @@ -204,9 +204,13 @@ static inline int is_shared_micin(struct snd_ac97 *ac97) /* The following snd_ac97_ymf753_... items added by David Shust (dshust@shustring.com) */ +/* Modified for YMF743 by Keita Maehara <maehara@debian.org> */ -/* It is possible to indicate to the Yamaha YMF753 the type of speakers being used. */ -static int snd_ac97_ymf753_info_speaker(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +/* It is possible to indicate to the Yamaha YMF7x3 the type of + speakers being used. */ + +static int snd_ac97_ymf7x3_info_speaker(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { static char *texts[3] = { "Standard", "Small", "Smaller" @@ -221,12 +225,13 @@ static int snd_ac97_ymf753_info_speaker(struct snd_kcontrol *kcontrol, struct sn return 0; } -static int snd_ac97_ymf753_get_speaker(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int snd_ac97_ymf7x3_get_speaker(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); unsigned short val; - val = ac97->regs[AC97_YMF753_3D_MODE_SEL]; + val = ac97->regs[AC97_YMF7X3_3D_MODE_SEL]; val = (val >> 10) & 3; if (val > 0) /* 0 = invalid */ val--; @@ -234,7 +239,8 @@ static int snd_ac97_ymf753_get_speaker(struct snd_kcontrol *kcontrol, struct snd return 0; } -static int snd_ac97_ymf753_put_speaker(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int snd_ac97_ymf7x3_put_speaker(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); unsigned short val; @@ -242,20 +248,22 @@ static int snd_ac97_ymf753_put_speaker(struct snd_kcontrol *kcontrol, struct snd if (ucontrol->value.enumerated.item[0] > 2) return -EINVAL; val = (ucontrol->value.enumerated.item[0] + 1) << 10; - return snd_ac97_update(ac97, AC97_YMF753_3D_MODE_SEL, val); + return snd_ac97_update(ac97, AC97_YMF7X3_3D_MODE_SEL, val); } -static const struct snd_kcontrol_new snd_ac97_ymf753_controls_speaker = +static const struct snd_kcontrol_new snd_ac97_ymf7x3_controls_speaker = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "3D Control - Speaker", - .info = snd_ac97_ymf753_info_speaker, - .get = snd_ac97_ymf753_get_speaker, - .put = snd_ac97_ymf753_put_speaker, + .info = snd_ac97_ymf7x3_info_speaker, + .get = snd_ac97_ymf7x3_get_speaker, + .put = snd_ac97_ymf7x3_put_speaker, }; -/* It is possible to indicate to the Yamaha YMF753 the source to direct to the S/PDIF output. */ -static int snd_ac97_ymf753_spdif_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +/* It is possible to indicate to the Yamaha YMF7x3 the source to + direct to the S/PDIF output. */ +static int snd_ac97_ymf7x3_spdif_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { static char *texts[2] = { "AC-Link", "A/D Converter" }; @@ -268,17 +276,19 @@ static int snd_ac97_ymf753_spdif_source_info(struct snd_kcontrol *kcontrol, stru return 0; } -static int snd_ac97_ymf753_spdif_source_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int snd_ac97_ymf7x3_spdif_source_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); unsigned short val; - val = ac97->regs[AC97_YMF753_DIT_CTRL2]; + val = ac97->regs[AC97_YMF7X3_DIT_CTRL]; ucontrol->value.enumerated.item[0] = (val >> 1) & 1; return 0; } -static int snd_ac97_ymf753_spdif_source_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int snd_ac97_ymf7x3_spdif_source_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); unsigned short val; @@ -286,7 +296,75 @@ static int snd_ac97_ymf753_spdif_source_put(struct snd_kcontrol *kcontrol, struc if (ucontrol->value.enumerated.item[0] > 1) return -EINVAL; val = ucontrol->value.enumerated.item[0] << 1; - return snd_ac97_update_bits(ac97, AC97_YMF753_DIT_CTRL2, 0x0002, val); + return snd_ac97_update_bits(ac97, AC97_YMF7X3_DIT_CTRL, 0x0002, val); +} + +static int patch_yamaha_ymf7x3_3d(struct snd_ac97 *ac97) +{ + struct snd_kcontrol *kctl; + int err; + + kctl = snd_ac97_cnew(&snd_ac97_controls_3d[0], ac97); + err = snd_ctl_add(ac97->bus->card, kctl); + if (err < 0) + return err; + strcpy(kctl->id.name, "3D Control - Wide"); + kctl->private_value = AC97_SINGLE_VALUE(AC97_3D_CONTROL, 9, 7, 0); + snd_ac97_write_cache(ac97, AC97_3D_CONTROL, 0x0000); + err = snd_ctl_add(ac97->bus->card, + snd_ac97_cnew(&snd_ac97_ymf7x3_controls_speaker, + ac97)); + if (err < 0) + return err; + snd_ac97_write_cache(ac97, AC97_YMF7X3_3D_MODE_SEL, 0x0c00); + return 0; +} + +static const struct snd_kcontrol_new snd_ac97_yamaha_ymf743_controls_spdif[3] = +{ + AC97_SINGLE(SNDRV_CTL_NAME_IEC958("", PLAYBACK, SWITCH), + AC97_YMF7X3_DIT_CTRL, 0, 1, 0), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, NONE) "Source", + .info = snd_ac97_ymf7x3_spdif_source_info, + .get = snd_ac97_ymf7x3_spdif_source_get, + .put = snd_ac97_ymf7x3_spdif_source_put, + }, + AC97_SINGLE(SNDRV_CTL_NAME_IEC958("", NONE, NONE) "Mute", + AC97_YMF7X3_DIT_CTRL, 2, 1, 1) +}; + +static int patch_yamaha_ymf743_build_spdif(struct snd_ac97 *ac97) +{ + int err; + + err = patch_build_controls(ac97, &snd_ac97_controls_spdif[0], 3); + if (err < 0) + return err; + err = patch_build_controls(ac97, + snd_ac97_yamaha_ymf743_controls_spdif, 3); + if (err < 0) + return err; + /* set default PCM S/PDIF params */ + /* PCM audio,no copyright,no preemphasis,PCM coder,original */ + snd_ac97_write_cache(ac97, AC97_YMF7X3_DIT_CTRL, 0xa201); + return 0; +} + +static struct snd_ac97_build_ops patch_yamaha_ymf743_ops = { + .build_spdif = patch_yamaha_ymf743_build_spdif, + .build_3d = patch_yamaha_ymf7x3_3d, +}; + +static int patch_yamaha_ymf743(struct snd_ac97 *ac97) +{ + ac97->build_ops = &patch_yamaha_ymf743_ops; + ac97->caps |= AC97_BC_BASS_TREBLE; + ac97->caps |= 0x04 << 10; /* Yamaha 3D enhancement */ + ac97->rates[AC97_RATES_SPDIF] = SNDRV_PCM_RATE_48000; /* 48k only */ + ac97->ext_id |= AC97_EI_SPDIF; /* force the detection of spdif */ + return 0; } /* The AC'97 spec states that the S/PDIF signal is to be output at pin 48. @@ -311,7 +389,7 @@ static int snd_ac97_ymf753_spdif_output_pin_get(struct snd_kcontrol *kcontrol, s struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); unsigned short val; - val = ac97->regs[AC97_YMF753_DIT_CTRL2]; + val = ac97->regs[AC97_YMF7X3_DIT_CTRL]; ucontrol->value.enumerated.item[0] = (val & 0x0008) ? 2 : (val & 0x0020) ? 1 : 0; return 0; } @@ -325,7 +403,7 @@ static int snd_ac97_ymf753_spdif_output_pin_put(struct snd_kcontrol *kcontrol, s return -EINVAL; val = (ucontrol->value.enumerated.item[0] == 2) ? 0x0008 : (ucontrol->value.enumerated.item[0] == 1) ? 0x0020 : 0; - return snd_ac97_update_bits(ac97, AC97_YMF753_DIT_CTRL2, 0x0028, val); + return snd_ac97_update_bits(ac97, AC97_YMF7X3_DIT_CTRL, 0x0028, val); /* The following can be used to direct S/PDIF output to pin 47 (EAPD). snd_ac97_write_cache(ac97, 0x62, snd_ac97_read(ac97, 0x62) | 0x0008); */ } @@ -334,9 +412,9 @@ static const struct snd_kcontrol_new snd_ac97_ymf753_controls_spdif[3] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", - .info = snd_ac97_ymf753_spdif_source_info, - .get = snd_ac97_ymf753_spdif_source_get, - .put = snd_ac97_ymf753_spdif_source_put, + .info = snd_ac97_ymf7x3_spdif_source_info, + .get = snd_ac97_ymf7x3_spdif_source_get, + .put = snd_ac97_ymf7x3_spdif_source_put, }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -345,25 +423,10 @@ static const struct snd_kcontrol_new snd_ac97_ymf753_controls_spdif[3] = { .get = snd_ac97_ymf753_spdif_output_pin_get, .put = snd_ac97_ymf753_spdif_output_pin_put, }, - AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",NONE,NONE) "Mute", AC97_YMF753_DIT_CTRL2, 2, 1, 1) + AC97_SINGLE(SNDRV_CTL_NAME_IEC958("", NONE, NONE) "Mute", + AC97_YMF7X3_DIT_CTRL, 2, 1, 1) }; -static int patch_yamaha_ymf753_3d(struct snd_ac97 * ac97) -{ - struct snd_kcontrol *kctl; - int err; - - if ((err = snd_ctl_add(ac97->bus->card, kctl = snd_ac97_cnew(&snd_ac97_controls_3d[0], ac97))) < 0) - return err; - strcpy(kctl->id.name, "3D Control - Wide"); - kctl->private_value = AC97_SINGLE_VALUE(AC97_3D_CONTROL, 9, 7, 0); - snd_ac97_write_cache(ac97, AC97_3D_CONTROL, 0x0000); - if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&snd_ac97_ymf753_controls_speaker, ac97))) < 0) - return err; - snd_ac97_write_cache(ac97, AC97_YMF753_3D_MODE_SEL, 0x0c00); - return 0; -} - static int patch_yamaha_ymf753_post_spdif(struct snd_ac97 * ac97) { int err; @@ -374,7 +437,7 @@ static int patch_yamaha_ymf753_post_spdif(struct snd_ac97 * ac97) } static struct snd_ac97_build_ops patch_yamaha_ymf753_ops = { - .build_3d = patch_yamaha_ymf753_3d, + .build_3d = patch_yamaha_ymf7x3_3d, .build_post_spdif = patch_yamaha_ymf753_post_spdif }; @@ -1880,14 +1943,7 @@ static int patch_ad1981b(struct snd_ac97 *ac97) return 0; } -static int snd_ac97_ad1888_lohpsel_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_ac97_ad1888_lohpsel_info snd_ctl_boolean_mono_info static int snd_ac97_ad1888_lohpsel_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -2186,15 +2242,7 @@ static int patch_ad1985(struct snd_ac97 * ac97) return 0; } -static int snd_ac97_ad1986_bool_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_ac97_ad1986_bool_info snd_ctl_boolean_mono_info static int snd_ac97_ad1986_lososel_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff --git a/sound/pci/ac97/ac97_patch.h b/sound/pci/ac97/ac97_patch.h index fd341ce63762..9cccc27ea1b5 100644 --- a/sound/pci/ac97/ac97_patch.h +++ b/sound/pci/ac97/ac97_patch.h @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Universal interface for Audio Codec '97 * * For more details look to AC '97 component specification revision 2.2 diff --git a/sound/pci/ac97/ac97_pcm.c b/sound/pci/ac97/ac97_pcm.c index 4281e6d0c5b6..8cbc03332b01 100644 --- a/sound/pci/ac97/ac97_pcm.c +++ b/sound/pci/ac97/ac97_pcm.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Universal interface for Audio Codec '97 * * For more details look to AC '97 component specification revision 2.2 diff --git a/sound/pci/ac97/ac97_proc.c b/sound/pci/ac97/ac97_proc.c index a3fdd7da911c..fed4a2c3d8a1 100644 --- a/sound/pci/ac97/ac97_proc.c +++ b/sound/pci/ac97/ac97_proc.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Universal interface for Audio Codec '97 * * For more details look to AC '97 component specification revision 2.2 @@ -236,10 +236,14 @@ static void snd_ac97_proc_read_main(struct snd_ac97 *ac97, struct snd_info_buffe val = snd_ac97_read(ac97, AC97_PCM_MIC_ADC_RATE); snd_iprintf(buffer, "PCM MIC ADC : %iHz\n", val); } - if ((ext & AC97_EI_SPDIF) || (ac97->flags & AC97_CS_SPDIF)) { + if ((ext & AC97_EI_SPDIF) || (ac97->flags & AC97_CS_SPDIF) || + (ac97->id == AC97_ID_YMF743)) { if (ac97->flags & AC97_CS_SPDIF) val = snd_ac97_read(ac97, AC97_CSR_SPDIF); - else + else if (ac97->id == AC97_ID_YMF743) { + val = snd_ac97_read(ac97, AC97_YMF7X3_DIT_CTRL); + val = 0x2000 | (val & 0xff00) >> 4 | (val & 0x38) >> 2; + } else val = snd_ac97_read(ac97, AC97_SPDIF); snd_iprintf(buffer, "SPDIF Control :%s%s%s%s Category=0x%x Generation=%i%s%s%s\n", diff --git a/sound/pci/ac97/ak4531_codec.c b/sound/pci/ac97/ak4531_codec.c index dc26820a03a5..722de451d15f 100644 --- a/sound/pci/ac97/ak4531_codec.c +++ b/sound/pci/ac97/ak4531_codec.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Universal routines for AK4531 codec * * @@ -29,7 +29,7 @@ #include <sound/ak4531_codec.h> #include <sound/tlv.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Universal routines for AK4531 codec"); MODULE_LICENSE("GPL"); diff --git a/sound/pci/ali5451/Makefile b/sound/pci/ali5451/Makefile index 2e1831597474..713459c12d22 100644 --- a/sound/pci/ali5451/Makefile +++ b/sound/pci/ali5451/Makefile @@ -1,6 +1,6 @@ # # Makefile for ALSA -# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz> +# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz> # snd-ali5451-objs := ali5451.o diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 05b4c8696941..4c2bd7adf674 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -1804,15 +1804,7 @@ static int __devinit snd_ali_build_pcms(struct snd_ali *codec) .info = snd_ali5451_spdif_info, .get = snd_ali5451_spdif_get, \ .put = snd_ali5451_spdif_put, .private_value = value} -static int snd_ali5451_spdif_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_ali5451_spdif_info snd_ctl_boolean_mono_info static int snd_ali5451_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index 8fb55d3b454b..1190ef366a41 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -1,7 +1,7 @@ /* * card-als4000.c - driver for Avance Logic ALS4000 based soundcards. * Copyright (C) 2000 by Bart Hartgers <bart@etpmod.phys.tue.nl>, - * Jaroslav Kysela <perex@suse.cz> + * Jaroslav Kysela <perex@perex.cz> * Copyright (C) 2002 by Andreas Mohr <hw7oshyuv3001@sneakemail.com> * * Framework borrowed from Massimo Piccioni's card-als100.c. diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c index 5ec1b6fcd548..f70286a7364a 100644 --- a/sound/pci/au88x0/au88x0.c +++ b/sound/pci/au88x0/au88x0.c @@ -232,6 +232,7 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip) pci_disable_device(chip->pci_dev); //FIXME: this not the right place to unregister the gameport vortex_gameport_unregister(chip); + kfree(chip); return err; } diff --git a/sound/pci/au88x0/au88x0_eq.c b/sound/pci/au88x0/au88x0_eq.c index 0c86a31c4336..38602b85874d 100644 --- a/sound/pci/au88x0/au88x0_eq.c +++ b/sound/pci/au88x0/au88x0_eq.c @@ -728,15 +728,7 @@ static void vortex_Eqlzr_shutdown(vortex_t * vortex) /* ALSA interface */ /* Control interface */ -static int -snd_vortex_eqtoggle_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_vortex_eqtoggle_info snd_ctl_boolean_mono_info static int snd_vortex_eqtoggle_get(struct snd_kcontrol *kcontrol, diff --git a/sound/pci/au88x0/au88x0_mpu401.c b/sound/pci/au88x0/au88x0_mpu401.c index c75d368ea087..8db3d3e6f7bb 100644 --- a/sound/pci/au88x0/au88x0_mpu401.c +++ b/sound/pci/au88x0/au88x0_mpu401.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Routines for control of MPU-401 in UART mode * * Modified for the Aureal Vortex based Soundcards diff --git a/sound/pci/au88x0/au88x0_synth.c b/sound/pci/au88x0/au88x0_synth.c index d3e662a1285d..978b856f5621 100644 --- a/sound/pci/au88x0/au88x0_synth.c +++ b/sound/pci/au88x0/au88x0_synth.c @@ -370,8 +370,8 @@ static void vortex_wt_SetFrequency(vortex_t * vortex, int wt, unsigned int sr) while ((edx & 0x80000000) == 0) { edx <<= 1; eax--; - if (eax == 0) ; - break; + if (eax == 0) + break; } if (eax) edx <<= 1; diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 131952f55857..91f9e6a112ff 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -147,15 +147,56 @@ MODULE_PARM_DESC(load_all, "Allow to load the non-whitelisted cards"); /* SYNC, one WRITE per line, one extra WRITE per page boundary, SYNC, JUMP */ #define MAX_RISC_SIZE ((1 + 255 + (PAGE_ALIGN(255 * 4092) / PAGE_SIZE - 1) + 1 + 1) * 8) +/* Cards with configuration information */ +enum snd_bt87x_boardid { + SND_BT87X_BOARD_UNKNOWN, + SND_BT87X_BOARD_GENERIC, /* both an & dig interfaces, 32kHz */ + SND_BT87X_BOARD_ANALOG, /* board with no external A/D */ + SND_BT87X_BOARD_OSPREY2x0, + SND_BT87X_BOARD_OSPREY440, + SND_BT87X_BOARD_AVPHONE98, +}; + +/* Card configuration */ +struct snd_bt87x_board { + int dig_rate; /* Digital input sampling rate */ + u32 digital_fmt; /* Register settings for digital input */ + unsigned no_analog:1; /* No analog input */ + unsigned no_digital:1; /* No digital input */ +}; + +static const __devinitdata struct snd_bt87x_board snd_bt87x_boards[] = { + [SND_BT87X_BOARD_UNKNOWN] = { + .dig_rate = 32000, /* just a guess */ + }, + [SND_BT87X_BOARD_GENERIC] = { + .dig_rate = 32000, + }, + [SND_BT87X_BOARD_ANALOG] = { + .no_digital = 1, + }, + [SND_BT87X_BOARD_OSPREY2x0] = { + .dig_rate = 44100, + .digital_fmt = CTL_DA_LRI | (1 << CTL_DA_LRD_SHIFT), + }, + [SND_BT87X_BOARD_OSPREY440] = { + .dig_rate = 32000, + .digital_fmt = CTL_DA_LRI | (1 << CTL_DA_LRD_SHIFT), + .no_analog = 1, + }, + [SND_BT87X_BOARD_AVPHONE98] = { + .dig_rate = 48000, + }, +}; + struct snd_bt87x { struct snd_card *card; struct pci_dev *pci; + struct snd_bt87x_board board; void __iomem *mmio; int irq; - int dig_rate; - spinlock_t reg_lock; unsigned long opened; struct snd_pcm_substream *substream; @@ -340,30 +381,11 @@ static struct snd_pcm_hardware snd_bt87x_analog_hw = { static int snd_bt87x_set_digital_hw(struct snd_bt87x *chip, struct snd_pcm_runtime *runtime) { - static struct { - int rate; - unsigned int bit; - } ratebits[] = { - {8000, SNDRV_PCM_RATE_8000}, - {11025, SNDRV_PCM_RATE_11025}, - {16000, SNDRV_PCM_RATE_16000}, - {22050, SNDRV_PCM_RATE_22050}, - {32000, SNDRV_PCM_RATE_32000}, - {44100, SNDRV_PCM_RATE_44100}, - {48000, SNDRV_PCM_RATE_48000} - }; - int i; - - chip->reg_control |= CTL_DA_IOM_DA; + chip->reg_control |= CTL_DA_IOM_DA | CTL_A_PWRDN; runtime->hw = snd_bt87x_digital_hw; - runtime->hw.rates = SNDRV_PCM_RATE_KNOT; - for (i = 0; i < ARRAY_SIZE(ratebits); ++i) - if (chip->dig_rate == ratebits[i].rate) { - runtime->hw.rates = ratebits[i].bit; - break; - } - runtime->hw.rate_min = chip->dig_rate; - runtime->hw.rate_max = chip->dig_rate; + runtime->hw.rates = snd_pcm_rate_to_rate_bit(chip->board.dig_rate); + runtime->hw.rate_min = chip->board.dig_rate; + runtime->hw.rate_max = chip->board.dig_rate; return 0; } @@ -380,7 +402,7 @@ static int snd_bt87x_set_analog_hw(struct snd_bt87x *chip, struct snd_pcm_runtim .rats = &analog_clock }; - chip->reg_control &= ~CTL_DA_IOM_DA; + chip->reg_control &= ~(CTL_DA_IOM_DA | CTL_A_PWRDN); runtime->hw = snd_bt87x_analog_hw; return snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraint_rates); @@ -419,6 +441,11 @@ static int snd_bt87x_close(struct snd_pcm_substream *substream) { struct snd_bt87x *chip = snd_pcm_substream_chip(substream); + spin_lock_irq(&chip->reg_lock); + chip->reg_control |= CTL_A_PWRDN; + snd_bt87x_writel(chip, REG_GPIO_DMA_CTL, chip->reg_control); + spin_unlock_irq(&chip->reg_lock); + chip->substream = NULL; clear_bit(0, &chip->opened); smp_mb__after_clear_bit(); @@ -569,15 +596,7 @@ static struct snd_kcontrol_new snd_bt87x_capture_volume = { .put = snd_bt87x_capture_volume_put, }; -static int snd_bt87x_capture_boost_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *info) -{ - info->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - info->count = 1; - info->value.integer.min = 0; - info->value.integer.max = 1; - return 0; -} +#define snd_bt87x_capture_boost_info snd_ctl_boolean_mono_info static int snd_bt87x_capture_boost_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *value) @@ -736,61 +755,69 @@ static int __devinit snd_bt87x_create(struct snd_card *card, chip->mmio = ioremap_nocache(pci_resource_start(pci, 0), pci_resource_len(pci, 0)); if (!chip->mmio) { - snd_bt87x_free(chip); snd_printk(KERN_ERR "cannot remap io memory\n"); - return -ENOMEM; + err = -ENOMEM; + goto fail; } - chip->reg_control = CTL_DA_ES2 | CTL_PKTP_16 | (15 << CTL_DA_SDR_SHIFT); + chip->reg_control = CTL_A_PWRDN | CTL_DA_ES2 | + CTL_PKTP_16 | (15 << CTL_DA_SDR_SHIFT); chip->interrupt_mask = MY_INTERRUPTS; snd_bt87x_writel(chip, REG_GPIO_DMA_CTL, chip->reg_control); snd_bt87x_writel(chip, REG_INT_MASK, 0); snd_bt87x_writel(chip, REG_INT_STAT, MY_INTERRUPTS); - if (request_irq(pci->irq, snd_bt87x_interrupt, IRQF_SHARED, - "Bt87x audio", chip)) { - snd_bt87x_free(chip); - snd_printk(KERN_ERR "cannot grab irq\n"); - return -EBUSY; + err = request_irq(pci->irq, snd_bt87x_interrupt, IRQF_SHARED, + "Bt87x audio", chip); + if (err < 0) { + snd_printk(KERN_ERR "cannot grab irq %d\n", pci->irq); + goto fail; } chip->irq = pci->irq; pci_set_master(pci); synchronize_irq(chip->irq); err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); - if (err < 0) { - snd_bt87x_free(chip); - return err; - } + if (err < 0) + goto fail; + snd_card_set_dev(card, &pci->dev); *rchip = chip; return 0; + +fail: + snd_bt87x_free(chip); + return err; } -#define BT_DEVICE(chip, subvend, subdev, rate) \ +#define BT_DEVICE(chip, subvend, subdev, id) \ { .vendor = PCI_VENDOR_ID_BROOKTREE, \ .device = chip, \ .subvendor = subvend, .subdevice = subdev, \ - .driver_data = rate } + .driver_data = SND_BT87X_BOARD_ ## id } +/* driver_data is the card id for that device */ -/* driver_data is the default digital_rate value for that device */ static struct pci_device_id snd_bt87x_ids[] = { /* Hauppauge WinTV series */ - BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0x13eb, 32000), + BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0x13eb, GENERIC), /* Hauppauge WinTV series */ - BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_879, 0x0070, 0x13eb, 32000), + BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_879, 0x0070, 0x13eb, GENERIC), /* Viewcast Osprey 200 */ - BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0xff01, 44100), + BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0xff01, OSPREY2x0), /* Viewcast Osprey 440 (rate is configurable via gpio) */ - BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0xff07, 32000), + BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0xff07, OSPREY440), /* ATI TV-Wonder */ - BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1002, 0x0001, 32000), + BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1002, 0x0001, GENERIC), /* Leadtek Winfast tv 2000xp delux */ - BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x107d, 0x6606, 32000), + BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x107d, 0x6606, GENERIC), /* Voodoo TV 200 */ - BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x121a, 0x3000, 32000), + BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x121a, 0x3000, GENERIC), /* AVerMedia Studio No. 103, 203, ...? */ - BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1461, 0x0003, 48000), + BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1461, 0x0003, AVPHONE98), + /* Prolink PixelView PV-M4900 */ + BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1554, 0x4011, GENERIC), + /* Pinnacle Studio PCTV rave */ + BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0xbd11, 0x1200, GENERIC), { } }; MODULE_DEVICE_TABLE(pci, snd_bt87x_ids); @@ -815,7 +842,7 @@ static struct { static struct pci_driver driver; -/* return the rate of the card, or a negative value if it's blacklisted */ +/* return the id of the card, or a negative value if it's blacklisted */ static int __devinit snd_bt87x_detect_card(struct pci_dev *pci) { int i; @@ -833,12 +860,12 @@ static int __devinit snd_bt87x_detect_card(struct pci_dev *pci) return -EBUSY; } - snd_printk(KERN_INFO "unknown card %#04x-%#04x:%#04x, using default rate 32000\n", - pci->device, pci->subsystem_vendor, pci->subsystem_device); + snd_printk(KERN_INFO "unknown card %#04x-%#04x:%#04x\n", + pci->device, pci->subsystem_vendor, pci->subsystem_device); snd_printk(KERN_DEBUG "please mail id, board name, and, " "if it works, the correct digital_rate option to " "<alsa-devel@alsa-project.org>\n"); - return 32000; /* default rate */ + return SND_BT87X_BOARD_UNKNOWN; } static int __devinit snd_bt87x_probe(struct pci_dev *pci, @@ -847,12 +874,16 @@ static int __devinit snd_bt87x_probe(struct pci_dev *pci, static int dev; struct snd_card *card; struct snd_bt87x *chip; - int err, rate; + int err; + enum snd_bt87x_boardid boardid; - rate = pci_id->driver_data; - if (! rate) - if ((rate = snd_bt87x_detect_card(pci)) <= 0) + if (!pci_id->driver_data) { + err = snd_bt87x_detect_card(pci); + if (err < 0) return -ENODEV; + boardid = err; + } else + boardid = pci_id->driver_data; if (dev >= SNDRV_CARDS) return -ENODEV; @@ -869,27 +900,39 @@ static int __devinit snd_bt87x_probe(struct pci_dev *pci, if (err < 0) goto _error; - if (digital_rate[dev] > 0) - chip->dig_rate = digital_rate[dev]; - else - chip->dig_rate = rate; + memcpy(&chip->board, &snd_bt87x_boards[boardid], sizeof(chip->board)); - err = snd_bt87x_pcm(chip, DEVICE_DIGITAL, "Bt87x Digital"); - if (err < 0) - goto _error; - err = snd_bt87x_pcm(chip, DEVICE_ANALOG, "Bt87x Analog"); - if (err < 0) - goto _error; + if (!chip->board.no_digital) { + if (digital_rate[dev] > 0) + chip->board.dig_rate = digital_rate[dev]; - err = snd_ctl_add(card, snd_ctl_new1(&snd_bt87x_capture_volume, chip)); - if (err < 0) - goto _error; - err = snd_ctl_add(card, snd_ctl_new1(&snd_bt87x_capture_boost, chip)); - if (err < 0) - goto _error; - err = snd_ctl_add(card, snd_ctl_new1(&snd_bt87x_capture_source, chip)); - if (err < 0) - goto _error; + chip->reg_control |= chip->board.digital_fmt; + + err = snd_bt87x_pcm(chip, DEVICE_DIGITAL, "Bt87x Digital"); + if (err < 0) + goto _error; + } + if (!chip->board.no_analog) { + err = snd_bt87x_pcm(chip, DEVICE_ANALOG, "Bt87x Analog"); + if (err < 0) + goto _error; + err = snd_ctl_add(card, snd_ctl_new1( + &snd_bt87x_capture_volume, chip)); + if (err < 0) + goto _error; + err = snd_ctl_add(card, snd_ctl_new1( + &snd_bt87x_capture_boost, chip)); + if (err < 0) + goto _error; + err = snd_ctl_add(card, snd_ctl_new1( + &snd_bt87x_capture_source, chip)); + if (err < 0) + goto _error; + } + snd_printk(KERN_INFO "bt87x%d: Using board %d, %sanalog, %sdigital " + "(rate %d Hz)\n", dev, boardid, + chip->board.no_analog ? "no " : "", + chip->board.no_digital ? "no " : "", chip->board.dig_rate); strcpy(card->driver, "Bt87x"); sprintf(card->shortname, "Brooktree Bt%x", pci->device); @@ -920,8 +963,8 @@ 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[] __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), + BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, PCI_ANY_ID, PCI_ANY_ID, UNKNOWN), + BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_879, PCI_ANY_ID, PCI_ANY_ID, UNKNOWN), { } }; diff --git a/sound/pci/ca0106/ca0106.h b/sound/pci/ca0106/ca0106.h index a0420bc63f0b..75da1746e758 100644 --- a/sound/pci/ca0106/ca0106.h +++ b/sound/pci/ca0106/ca0106.h @@ -1,7 +1,7 @@ /* * Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk> * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit - * Version: 0.0.21 + * Version: 0.0.22 * * FEATURES currently supported: * See ca0106_main.c for features. @@ -47,6 +47,8 @@ * Added GPIO info for SB Live 24bit. * 0.0.21 * Implement support for Line-in capture on SB Live 24bit. + * 0.0.22 + * Add support for mute control on SB Live 24bit (cards w/ SPI DAC) * * * This code was initally based on code from ALSA's emu10k1x.c which is: @@ -552,6 +554,95 @@ #define CONTROL_CENTER_LFE_CHANNEL 1 #define CONTROL_UNKNOWN_CHANNEL 2 + +/* Based on WM8768 Datasheet Rev 4.2 page 32 */ +#define SPI_REG_MASK 0x1ff /* 16-bit SPI writes have a 7-bit address */ +#define SPI_REG_SHIFT 9 /* followed by 9 bits of data */ + +#define SPI_LDA1_REG 0 /* digital attenuation */ +#define SPI_RDA1_REG 1 +#define SPI_LDA2_REG 4 +#define SPI_RDA2_REG 5 +#define SPI_LDA3_REG 6 +#define SPI_RDA3_REG 7 +#define SPI_LDA4_REG 13 +#define SPI_RDA4_REG 14 +#define SPI_MASTDA_REG 8 + +#define SPI_DA_BIT_UPDATE (1<<8) /* update attenuation values */ +#define SPI_DA_BIT_0dB 0xff /* 0 dB */ +#define SPI_DA_BIT_infdB 0x00 /* inf dB attenuation (mute) */ + +#define SPI_PL_REG 2 +#define SPI_PL_BIT_L_M (0<<5) /* left channel = mute */ +#define SPI_PL_BIT_L_L (1<<5) /* left channel = left */ +#define SPI_PL_BIT_L_R (2<<5) /* left channel = right */ +#define SPI_PL_BIT_L_C (3<<5) /* left channel = (L+R)/2 */ +#define SPI_PL_BIT_R_M (0<<7) /* right channel = mute */ +#define SPI_PL_BIT_R_L (1<<7) /* right channel = left */ +#define SPI_PL_BIT_R_R (2<<7) /* right channel = right */ +#define SPI_PL_BIT_R_C (3<<7) /* right channel = (L+R)/2 */ +#define SPI_IZD_REG 2 +#define SPI_IZD_BIT (1<<4) /* infinite zero detect */ + +#define SPI_FMT_REG 3 +#define SPI_FMT_BIT_RJ (0<<0) /* right justified mode */ +#define SPI_FMT_BIT_LJ (1<<0) /* left justified mode */ +#define SPI_FMT_BIT_I2S (2<<0) /* I2S mode */ +#define SPI_FMT_BIT_DSP (3<<0) /* DSP Modes A or B */ +#define SPI_LRP_REG 3 +#define SPI_LRP_BIT (1<<2) /* invert LRCLK polarity */ +#define SPI_BCP_REG 3 +#define SPI_BCP_BIT (1<<3) /* invert BCLK polarity */ +#define SPI_IWL_REG 3 +#define SPI_IWL_BIT_16 (0<<4) /* 16-bit world length */ +#define SPI_IWL_BIT_20 (1<<4) /* 20-bit world length */ +#define SPI_IWL_BIT_24 (2<<4) /* 24-bit world length */ +#define SPI_IWL_BIT_32 (3<<4) /* 32-bit world length */ + +#define SPI_MS_REG 10 +#define SPI_MS_BIT (1<<5) /* master mode */ +#define SPI_RATE_REG 10 /* only applies in master mode */ +#define SPI_RATE_BIT_128 (0<<6) /* MCLK = LRCLK * 128 */ +#define SPI_RATE_BIT_192 (1<<6) +#define SPI_RATE_BIT_256 (2<<6) +#define SPI_RATE_BIT_384 (3<<6) +#define SPI_RATE_BIT_512 (4<<6) +#define SPI_RATE_BIT_768 (5<<6) + +/* They really do label the bit for the 4th channel "4" and not "3" */ +#define SPI_DMUTE0_REG 9 +#define SPI_DMUTE1_REG 9 +#define SPI_DMUTE2_REG 9 +#define SPI_DMUTE4_REG 15 +#define SPI_DMUTE0_BIT (1<<3) +#define SPI_DMUTE1_BIT (1<<4) +#define SPI_DMUTE2_BIT (1<<5) +#define SPI_DMUTE4_BIT (1<<2) + +#define SPI_PHASE0_REG 3 +#define SPI_PHASE1_REG 3 +#define SPI_PHASE2_REG 3 +#define SPI_PHASE4_REG 15 +#define SPI_PHASE0_BIT (1<<6) +#define SPI_PHASE1_BIT (1<<7) +#define SPI_PHASE2_BIT (1<<8) +#define SPI_PHASE4_BIT (1<<3) + +#define SPI_PDWN_REG 2 /* power down all DACs */ +#define SPI_PDWN_BIT (1<<2) +#define SPI_DACD0_REG 10 /* power down individual DACs */ +#define SPI_DACD1_REG 10 +#define SPI_DACD2_REG 10 +#define SPI_DACD4_REG 15 +#define SPI_DACD0_BIT (1<<1) +#define SPI_DACD1_BIT (1<<2) +#define SPI_DACD2_BIT (1<<3) +#define SPI_DACD4_BIT (1<<0) /* datasheet error says it's 1 */ + +#define SPI_PWRDNALL_REG 10 /* power down everything */ +#define SPI_PWRDNALL_BIT (1<<4) + #include "ca_midi.h" struct snd_ca0106; @@ -611,6 +702,8 @@ struct snd_ca0106 { struct snd_ca_midi midi; struct snd_ca_midi midi2; + + u16 spi_dac_reg[16]; }; int snd_ca0106_mixer(struct snd_ca0106 *emu); @@ -627,4 +720,5 @@ void snd_ca0106_ptr_write(struct snd_ca0106 *emu, int snd_ca0106_i2c_write(struct snd_ca0106 *emu, u32 reg, u32 value); - +int snd_ca0106_spi_write(struct snd_ca0106 * emu, + unsigned int data); diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index fcab8fb97e38..31d8db9f7a4c 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -1,7 +1,7 @@ /* * Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk> * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit - * Version: 0.0.23 + * Version: 0.0.25 * * FEATURES currently supported: * Front, Rear and Center/LFE. @@ -79,6 +79,10 @@ * Add support for MSI K8N Diamond Motherboard with onboard SB Live 24bit without AC97. From kiksen, bug #901 * 0.0.23 * Implement support for Line-in capture on SB Live 24bit. + * 0.0.24 + * Add support for mute control on SB Live 24bit (cards w/ SPI DAC) + * 0.0.25 + * Powerdown SPI DAC channels when not in use * * BUGS: * Some stability problems when unloading the snd-ca0106 kernel module. @@ -170,6 +174,15 @@ MODULE_PARM_DESC(subsystem, "Force card subsystem model."); static struct snd_ca0106_details ca0106_chip_details[] = { /* Sound Blaster X-Fi Extreme Audio. This does not have an AC97. 53SB079000000 */ /* It is really just a normal SB Live 24bit. */ + /* Tested: + * See ALSA bug#3251 + */ + { .serial = 0x10131102, + .name = "X-Fi Extreme Audio [SBxxxx]", + .gpio_type = 1, + .i2c_adc = 1 } , + /* Sound Blaster X-Fi Extreme Audio. This does not have an AC97. 53SB079000000 */ + /* It is really just a normal SB Live 24bit. */ /* * CTRL:CA0111-WTLF * ADC: WM8775SEDS @@ -261,10 +274,11 @@ static struct snd_ca0106_details ca0106_chip_details[] = { /* hardware definition */ static struct snd_pcm_hardware snd_ca0106_playback_hw = { - .info = (SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP_VALID), + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_SYNC_START, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, .rates = (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000), @@ -447,6 +461,19 @@ static void snd_ca0106_pcm_free_substream(struct snd_pcm_runtime *runtime) kfree(runtime->private_data); } +static const int spi_dacd_reg[] = { + [PCM_FRONT_CHANNEL] = SPI_DACD4_REG, + [PCM_REAR_CHANNEL] = SPI_DACD0_REG, + [PCM_CENTER_LFE_CHANNEL]= SPI_DACD2_REG, + [PCM_UNKNOWN_CHANNEL] = SPI_DACD1_REG, +}; +static const int spi_dacd_bit[] = { + [PCM_FRONT_CHANNEL] = SPI_DACD4_BIT, + [PCM_REAR_CHANNEL] = SPI_DACD0_BIT, + [PCM_CENTER_LFE_CHANNEL]= SPI_DACD2_BIT, + [PCM_UNKNOWN_CHANNEL] = SPI_DACD1_BIT, +}; + /* open_playback callback */ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substream, int channel_id) @@ -481,6 +508,17 @@ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substr return err; if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0) return err; + snd_pcm_set_sync(substream); + + if (chip->details->spi_dac && channel_id != PCM_FRONT_CHANNEL) { + const int reg = spi_dacd_reg[channel_id]; + + /* Power up dac */ + chip->spi_dac_reg[reg] &= ~spi_dacd_bit[channel_id]; + err = snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]); + if (err < 0) + return err; + } return 0; } @@ -491,6 +529,14 @@ static int snd_ca0106_pcm_close_playback(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct snd_ca0106_pcm *epcm = runtime->private_data; chip->playback_channels[epcm->channel_id].use = 0; + + if (chip->details->spi_dac && epcm->channel_id != PCM_FRONT_CHANNEL) { + const int reg = spi_dacd_reg[epcm->channel_id]; + + /* Power down DAC */ + chip->spi_dac_reg[reg] |= spi_dacd_bit[epcm->channel_id]; + snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]); + } /* FIXME: maybe zero others */ return 0; } @@ -809,6 +855,9 @@ static int snd_ca0106_pcm_trigger_playback(struct snd_pcm_substream *substream, break; } snd_pcm_group_for_each_entry(s, substream) { + if (snd_pcm_substream_chip(s) != emu || + s->stream != SNDRV_PCM_STREAM_PLAYBACK) + continue; runtime = s->runtime; epcm = runtime->private_data; channel = epcm->channel_id; @@ -1214,28 +1263,23 @@ static int __devinit snd_ca0106_pcm(struct snd_ca0106 *emu, int device, struct s return 0; } +#define SPI_REG(reg, value) (((reg) << SPI_REG_SHIFT) | (value)) static unsigned int spi_dac_init[] = { - 0x00ff, - 0x02ff, - 0x0400, - 0x0520, - 0x0620, /* Set 24 bit. Was 0x0600 */ - 0x08ff, - 0x0aff, - 0x0cff, - 0x0eff, - 0x10ff, - 0x1200, - 0x1400, - 0x1480, - 0x1800, - 0x1aff, - 0x1cff, - 0x1e00, - 0x0530, - 0x0602, - 0x0622, - 0x1400, + SPI_REG(SPI_LDA1_REG, SPI_DA_BIT_0dB), /* 0dB dig. attenuation */ + SPI_REG(SPI_RDA1_REG, SPI_DA_BIT_0dB), + SPI_REG(SPI_PL_REG, SPI_PL_BIT_L_L | SPI_PL_BIT_R_R | SPI_IZD_BIT), + SPI_REG(SPI_FMT_REG, SPI_FMT_BIT_I2S | SPI_IWL_BIT_24), + SPI_REG(SPI_LDA2_REG, SPI_DA_BIT_0dB), + SPI_REG(SPI_RDA2_REG, SPI_DA_BIT_0dB), + SPI_REG(SPI_LDA3_REG, SPI_DA_BIT_0dB), + SPI_REG(SPI_RDA3_REG, SPI_DA_BIT_0dB), + SPI_REG(SPI_MASTDA_REG, SPI_DA_BIT_0dB), + SPI_REG(9, 0x00), + SPI_REG(SPI_MS_REG, SPI_DACD0_BIT | SPI_DACD1_BIT | SPI_DACD2_BIT), + SPI_REG(12, 0x00), + SPI_REG(SPI_LDA4_REG, SPI_DA_BIT_0dB), + SPI_REG(SPI_RDA4_REG, SPI_DA_BIT_0dB | SPI_DA_BIT_UPDATE), + SPI_REG(SPI_DACD4_REG, 0x00), }; static unsigned int i2c_adc_init[][2] = { @@ -1475,8 +1519,13 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, int size, n; size = ARRAY_SIZE(spi_dac_init); - for (n=0; n < size; n++) + for (n = 0; n < size; n++) { + int reg = spi_dac_init[n] >> SPI_REG_SHIFT; + snd_ca0106_spi_write(chip, spi_dac_init[n]); + if (reg < ARRAY_SIZE(chip->spi_dac_reg)) + chip->spi_dac_reg[reg] = spi_dac_init[n]; + } } if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index 9c3a9c8d1dc2..be519a17dfa5 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -1,7 +1,7 @@ /* * Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk> * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit - * Version: 0.0.17 + * Version: 0.0.18 * * FEATURES currently supported: * See ca0106_main.c for features. @@ -39,6 +39,8 @@ * Modified Copyright message. * 0.0.17 * Implement Mic and Line in Capture. + * 0.0.18 + * Add support for mute control on SB Live 24bit (cards w/ SPI DAC) * * This code was initally based on code from ALSA's emu10k1x.c which is: * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com> @@ -77,15 +79,7 @@ static const DECLARE_TLV_DB_SCALE(snd_ca0106_db_scale1, -5175, 25, 1); static const DECLARE_TLV_DB_SCALE(snd_ca0106_db_scale2, -10350, 50, 1); -static int snd_ca0106_shared_spdif_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_ca0106_shared_spdif_info snd_ctl_boolean_mono_info static int snd_ca0106_shared_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -470,6 +464,42 @@ static int snd_ca0106_i2c_volume_put(struct snd_kcontrol *kcontrol, return change; } +#define spi_mute_info snd_ctl_boolean_mono_info + +static int spi_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + unsigned int reg = kcontrol->private_value >> SPI_REG_SHIFT; + unsigned int bit = kcontrol->private_value & SPI_REG_MASK; + + ucontrol->value.integer.value[0] = !(emu->spi_dac_reg[reg] & bit); + return 0; +} + +static int spi_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + unsigned int reg = kcontrol->private_value >> SPI_REG_SHIFT; + unsigned int bit = kcontrol->private_value & SPI_REG_MASK; + int ret; + + ret = emu->spi_dac_reg[reg] & bit; + if (ucontrol->value.integer.value[0]) { + if (!ret) /* bit already cleared, do nothing */ + return 0; + emu->spi_dac_reg[reg] &= ~bit; + } else { + if (ret) /* bit already set, do nothing */ + return 0; + emu->spi_dac_reg[reg] |= bit; + } + + ret = snd_ca0106_spi_write(emu, emu->spi_dac_reg[reg]); + return ret ? -1 : 1; +} + #define CA_VOLUME(xname,chid,reg) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ @@ -562,6 +592,28 @@ static struct snd_kcontrol_new snd_ca0106_volume_i2c_adc_ctls[] __devinitdata = I2C_VOLUME("Aux Capture Volume", 3), }; +#define SPI_SWITCH(xname,reg,bit) \ +{ \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .info = spi_mute_info, \ + .get = spi_mute_get, \ + .put = spi_mute_put, \ + .private_value = (reg<<SPI_REG_SHIFT) | (bit) \ +} + +static struct snd_kcontrol_new snd_ca0106_volume_spi_dac_ctls[] +__devinitdata = { + SPI_SWITCH("Analog Front Playback Switch", + SPI_DMUTE4_REG, SPI_DMUTE4_BIT), + SPI_SWITCH("Analog Rear Playback Switch", + SPI_DMUTE0_REG, SPI_DMUTE0_BIT), + SPI_SWITCH("Analog Center/LFE Playback Switch", + SPI_DMUTE2_REG, SPI_DMUTE2_BIT), + SPI_SWITCH("Analog Side Playback Switch", + SPI_DMUTE1_REG, SPI_DMUTE1_BIT), +}; + static int __devinit remove_ctl(struct snd_card *card, const char *name) { struct snd_ctl_elem_id id; @@ -591,9 +643,19 @@ static int __devinit rename_ctl(struct snd_card *card, const char *src, const ch return -ENOENT; } +#define ADD_CTLS(emu, ctls) \ + do { \ + int i, err; \ + for (i = 0; i < ARRAY_SIZE(ctls); i++) { \ + err = snd_ctl_add(card, snd_ctl_new1(&ctls[i], emu)); \ + if (err < 0) \ + return err; \ + } \ + } while (0) + int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu) { - int i, err; + int err; struct snd_card *card = emu->card; char **c; static char *ca0106_remove_ctls[] = { @@ -640,17 +702,9 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu) rename_ctl(card, c[0], c[1]); #endif - for (i = 0; i < ARRAY_SIZE(snd_ca0106_volume_ctls); i++) { - err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_volume_ctls[i], emu)); - if (err < 0) - return err; - } + ADD_CTLS(emu, snd_ca0106_volume_ctls); if (emu->details->i2c_adc == 1) { - for (i = 0; i < ARRAY_SIZE(snd_ca0106_volume_i2c_adc_ctls); i++) { - err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_volume_i2c_adc_ctls[i], emu)); - if (err < 0) - return err; - } + ADD_CTLS(emu, snd_ca0106_volume_i2c_adc_ctls); if (emu->details->gpio_type == 1) err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_capture_mic_line_in, emu)); else /* gpio_type == 2 */ @@ -658,6 +712,8 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu) if (err < 0) return err; } + if (emu->details->spi_dac == 1) + ADD_CTLS(emu, snd_ca0106_volume_spi_dac_ctls); return 0; } diff --git a/sound/pci/ca0106/ca_midi.c b/sound/pci/ca0106/ca_midi.c index 2e6eab1f1189..ad32eff2713f 100644 --- a/sound/pci/ca0106/ca_midi.c +++ b/sound/pci/ca0106/ca_midi.c @@ -6,7 +6,7 @@ * Changelog: * Implementation is based on mpu401 and emu10k1x and * tested with ca0106. - * mpu401: Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * mpu401: Copyright (c) by Jaroslav Kysela <perex@perex.cz> * emu10k1x: Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com> * * This program is free software; you can redistribute it and/or modify diff --git a/sound/pci/ca0106/ca_midi.h b/sound/pci/ca0106/ca_midi.h index b72c0933bd22..922ed3e3731e 100644 --- a/sound/pci/ca0106/ca_midi.h +++ b/sound/pci/ca0106/ca_midi.h @@ -22,9 +22,9 @@ * */ -#include<linux/spinlock.h> -#include<sound/rawmidi.h> -#include<sound/mpu401.h> +#include <linux/spinlock.h> +#include <sound/rawmidi.h> +#include <sound/mpu401.h> #define CA_MIDI_MODE_INPUT MPU401_MODE_INPUT #define CA_MIDI_MODE_OUTPUT MPU401_MODE_OUTPUT diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 7d3c5ee0005c..6832649879ce 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -95,30 +95,34 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address."); #define CM_CHADC0 0x00000001 /* ch0, 0:playback, 1:record */ #define CM_REG_FUNCTRL1 0x04 -#define CM_ASFC_MASK 0x0000E000 /* ADC sampling frequency */ -#define CM_ASFC_SHIFT 13 -#define CM_DSFC_MASK 0x00001C00 /* DAC sampling frequency */ -#define CM_DSFC_SHIFT 10 +#define CM_DSFC_MASK 0x0000E000 /* channel 1 (DAC?) sampling frequency */ +#define CM_DSFC_SHIFT 13 +#define CM_ASFC_MASK 0x00001C00 /* channel 0 (ADC?) sampling frequency */ +#define CM_ASFC_SHIFT 10 #define CM_SPDF_1 0x00000200 /* SPDIF IN/OUT at channel B */ #define CM_SPDF_0 0x00000100 /* SPDIF OUT only channel A */ -#define CM_SPDFLOOP 0x00000080 /* ext. SPDIIF/OUT -> IN loopback */ +#define CM_SPDFLOOP 0x00000080 /* ext. SPDIIF/IN -> OUT loopback */ #define CM_SPDO2DAC 0x00000040 /* SPDIF/OUT can be heard from internal DAC */ #define CM_INTRM 0x00000020 /* master control block (MCB) interrupt enabled */ #define CM_BREQ 0x00000010 /* bus master enabled */ #define CM_VOICE_EN 0x00000008 /* legacy voice (SB16,FM) */ -#define CM_UART_EN 0x00000004 /* UART */ -#define CM_JYSTK_EN 0x00000002 /* joy stick */ +#define CM_UART_EN 0x00000004 /* legacy UART */ +#define CM_JYSTK_EN 0x00000002 /* legacy joystick */ +#define CM_ZVPORT 0x00000001 /* ZVPORT */ #define CM_REG_CHFORMAT 0x08 #define CM_CHB3D5C 0x80000000 /* 5,6 channels */ +#define CM_FMOFFSET2 0x40000000 /* initial FM PCM offset 2 when Fmute=1 */ #define CM_CHB3D 0x20000000 /* 4 channels */ #define CM_CHIP_MASK1 0x1f000000 #define CM_CHIP_037 0x01000000 - -#define CM_SPDIF_SELECT1 0x00080000 /* for model <= 037 ? */ +#define CM_SETLAT48 0x00800000 /* set latency timer 48h */ +#define CM_EDGEIRQ 0x00400000 /* emulated edge trigger legacy IRQ */ +#define CM_SPD24SEL39 0x00200000 /* 24-bit spdif: model 039 */ #define CM_AC3EN1 0x00100000 /* enable AC3: model 037 */ +#define CM_SPDIF_SELECT1 0x00080000 /* for model <= 037 ? */ #define CM_SPD24SEL 0x00020000 /* 24bit spdif: model 037 */ /* #define CM_SPDIF_INVERSE 0x00010000 */ /* ??? */ @@ -128,35 +132,45 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address."); #define CM_ADCBITLEN_14 0x00008000 #define CM_ADCBITLEN_13 0x0000C000 -#define CM_ADCDACLEN_MASK 0x00003000 +#define CM_ADCDACLEN_MASK 0x00003000 /* model 037 */ #define CM_ADCDACLEN_060 0x00000000 #define CM_ADCDACLEN_066 0x00001000 #define CM_ADCDACLEN_130 0x00002000 #define CM_ADCDACLEN_280 0x00003000 +#define CM_ADCDLEN_MASK 0x00003000 /* model 039 */ +#define CM_ADCDLEN_ORIGINAL 0x00000000 +#define CM_ADCDLEN_EXTRA 0x00001000 +#define CM_ADCDLEN_24K 0x00002000 +#define CM_ADCDLEN_WEIGHT 0x00003000 + #define CM_CH1_SRATE_176K 0x00000800 +#define CM_CH1_SRATE_96K 0x00000800 /* model 055? */ #define CM_CH1_SRATE_88K 0x00000400 #define CM_CH0_SRATE_176K 0x00000200 +#define CM_CH0_SRATE_96K 0x00000200 /* model 055? */ #define CM_CH0_SRATE_88K 0x00000100 #define CM_SPDIF_INVERSE2 0x00000080 /* model 055? */ +#define CM_DBLSPDS 0x00000040 /* double SPDIF sample rate 88.2/96 */ +#define CM_POLVALID 0x00000020 /* inverse SPDIF/IN valid bit */ +#define CM_SPDLOCKED 0x00000010 -#define CM_CH1FMT_MASK 0x0000000C +#define CM_CH1FMT_MASK 0x0000000C /* bit 3: 16 bits, bit 2: stereo */ #define CM_CH1FMT_SHIFT 2 -#define CM_CH0FMT_MASK 0x00000003 +#define CM_CH0FMT_MASK 0x00000003 /* bit 1: 16 bits, bit 0: stereo */ #define CM_CH0FMT_SHIFT 0 #define CM_REG_INT_HLDCLR 0x0C #define CM_CHIP_MASK2 0xff000000 +#define CM_CHIP_8768 0x20000000 +#define CM_CHIP_055 0x08000000 #define CM_CHIP_039 0x04000000 #define CM_CHIP_039_6CH 0x01000000 -#define CM_CHIP_055 0x08000000 -#define CM_CHIP_8768 0x20000000 +#define CM_UNKNOWN_INT_EN 0x00080000 /* ? */ #define CM_TDMA_INT_EN 0x00040000 #define CM_CH1_INT_EN 0x00020000 #define CM_CH0_INT_EN 0x00010000 -#define CM_INT_HOLD 0x00000002 -#define CM_INT_CLEAR 0x00000001 #define CM_REG_INT_STATUS 0x10 #define CM_INTR 0x80000000 @@ -175,12 +189,13 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address."); #define CM_CHINT0 0x00000001 #define CM_REG_LEGACY_CTRL 0x14 -#define CM_NXCHG 0x80000000 /* h/w multi channels? */ +#define CM_NXCHG 0x80000000 /* don't map base reg dword->sample */ #define CM_VMPU_MASK 0x60000000 /* MPU401 i/o port address */ #define CM_VMPU_330 0x00000000 #define CM_VMPU_320 0x20000000 #define CM_VMPU_310 0x40000000 #define CM_VMPU_300 0x60000000 +#define CM_ENWR8237 0x10000000 /* enable bus master to write 8237 base reg */ #define CM_VSBSEL_MASK 0x0C000000 /* SB16 base address */ #define CM_VSBSEL_220 0x00000000 #define CM_VSBSEL_240 0x04000000 @@ -191,44 +206,74 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address."); #define CM_FMSEL_3C8 0x01000000 #define CM_FMSEL_3E0 0x02000000 #define CM_FMSEL_3E8 0x03000000 -#define CM_ENSPDOUT 0x00800000 /* enable XPDIF/OUT to I/O interface */ -#define CM_SPDCOPYRHT 0x00400000 /* set copyright spdif in/out */ +#define CM_ENSPDOUT 0x00800000 /* enable XSPDIF/OUT to I/O interface */ +#define CM_SPDCOPYRHT 0x00400000 /* spdif in/out copyright bit */ #define CM_DAC2SPDO 0x00200000 /* enable wave+fm_midi -> SPDIF/OUT */ -#define CM_SETRETRY 0x00010000 /* 0: legacy i/o wait (default), 1: legacy i/o bus retry */ +#define CM_INVIDWEN 0x00100000 /* internal vendor ID write enable, model 039? */ +#define CM_SETRETRY 0x00100000 /* 0: legacy i/o wait (default), 1: legacy i/o bus retry */ +#define CM_C_EEACCESS 0x00080000 /* direct programming eeprom regs */ +#define CM_C_EECS 0x00040000 +#define CM_C_EEDI46 0x00020000 +#define CM_C_EECK46 0x00010000 #define CM_CHB3D6C 0x00008000 /* 5.1 channels support */ -#define CM_LINE_AS_BASS 0x00006000 /* use line-in as bass */ +#define CM_CENTR2LIN 0x00004000 /* line-in as center out */ +#define CM_BASE2LIN 0x00002000 /* line-in as bass out */ +#define CM_EXBASEN 0x00001000 /* external bass input enable */ #define CM_REG_MISC_CTRL 0x18 -#define CM_PWD 0x80000000 +#define CM_PWD 0x80000000 /* power down */ #define CM_RESET 0x40000000 -#define CM_SFIL_MASK 0x30000000 -#define CM_TXVX 0x08000000 -#define CM_N4SPK3D 0x04000000 /* 4ch output */ +#define CM_SFIL_MASK 0x30000000 /* filter control at front end DAC, model 037? */ +#define CM_VMGAIN 0x10000000 /* analog master amp +6dB, model 039? */ +#define CM_TXVX 0x08000000 /* model 037? */ +#define CM_N4SPK3D 0x04000000 /* copy front to rear */ #define CM_SPDO5V 0x02000000 /* 5V spdif output (1 = 0.5v (coax)) */ #define CM_SPDIF48K 0x01000000 /* write */ #define CM_SPATUS48K 0x01000000 /* read */ -#define CM_ENDBDAC 0x00800000 /* enable dual dac */ +#define CM_ENDBDAC 0x00800000 /* enable double dac */ #define CM_XCHGDAC 0x00400000 /* 0: front=ch0, 1: front=ch1 */ #define CM_SPD32SEL 0x00200000 /* 0: 16bit SPDIF, 1: 32bit */ -#define CM_SPDFLOOPI 0x00100000 /* int. SPDIF-IN -> int. OUT */ -#define CM_FM_EN 0x00080000 /* enalbe FM */ +#define CM_SPDFLOOPI 0x00100000 /* int. SPDIF-OUT -> int. IN */ +#define CM_FM_EN 0x00080000 /* enable legacy FM */ #define CM_AC3EN2 0x00040000 /* enable AC3: model 039 */ -#define CM_VIDWPDSB 0x00010000 +#define CM_ENWRASID 0x00010000 /* choose writable internal SUBID (audio) */ +#define CM_VIDWPDSB 0x00010000 /* model 037? */ #define CM_SPDF_AC97 0x00008000 /* 0: SPDIF/OUT 44.1K, 1: 48K */ -#define CM_MASK_EN 0x00004000 -#define CM_VIDWPPRT 0x00002000 -#define CM_SFILENB 0x00001000 -#define CM_MMODE_MASK 0x00000E00 +#define CM_MASK_EN 0x00004000 /* activate channel mask on legacy DMA */ +#define CM_ENWRMSID 0x00002000 /* choose writable internal SUBID (modem) */ +#define CM_VIDWPPRT 0x00002000 /* model 037? */ +#define CM_SFILENB 0x00001000 /* filter stepping at front end DAC, model 037? */ +#define CM_MMODE_MASK 0x00000E00 /* model DAA interface mode */ #define CM_SPDIF_SELECT2 0x00000100 /* for model > 039 ? */ #define CM_ENCENTER 0x00000080 -#define CM_FLINKON 0x00000040 -#define CM_FLINKOFF 0x00000020 -#define CM_MIDSMP 0x00000010 -#define CM_UPDDMA_MASK 0x0000000C -#define CM_TWAIT_MASK 0x00000003 +#define CM_FLINKON 0x00000080 /* force modem link detection on, model 037 */ +#define CM_MUTECH1 0x00000040 /* mute PCI ch1 to DAC */ +#define CM_FLINKOFF 0x00000040 /* force modem link detection off, model 037 */ +#define CM_UNKNOWN_18_5 0x00000020 /* ? */ +#define CM_MIDSMP 0x00000010 /* 1/2 interpolation at front end DAC */ +#define CM_UPDDMA_MASK 0x0000000C /* TDMA position update notification */ +#define CM_UPDDMA_2048 0x00000000 +#define CM_UPDDMA_1024 0x00000004 +#define CM_UPDDMA_512 0x00000008 +#define CM_UPDDMA_256 0x0000000C +#define CM_TWAIT_MASK 0x00000003 /* model 037 */ +#define CM_TWAIT1 0x00000002 /* FM i/o cycle, 0: 48, 1: 64 PCICLKs */ +#define CM_TWAIT0 0x00000001 /* i/o cycle, 0: 4, 1: 6 PCICLKs */ + +#define CM_REG_TDMA_POSITION 0x1C +#define CM_TDMA_CNT_MASK 0xFFFF0000 /* current byte/word count */ +#define CM_TDMA_ADR_MASK 0x0000FFFF /* current address */ /* byte */ #define CM_REG_MIXER0 0x20 +#define CM_REG_SBVR 0x20 /* write: sb16 version */ +#define CM_REG_DEV 0x20 /* read: hardware device version */ + +#define CM_REG_MIXER21 0x21 +#define CM_UNKNOWN_21_MASK 0x78 /* ? */ +#define CM_X_ADPCM 0x04 /* SB16 ADPCM enable */ +#define CM_PROINV 0x02 /* SBPro left/right channel switching */ +#define CM_X_SB16 0x01 /* SB16 compatible */ #define CM_REG_SB16_DATA 0x22 #define CM_REG_SB16_ADDR 0x23 @@ -243,8 +288,8 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address."); #define CM_FMMUTE_SHIFT 7 #define CM_WSMUTE 0x40 /* mute PCM */ #define CM_WSMUTE_SHIFT 6 -#define CM_SPK4 0x20 /* lin-in -> rear line out */ -#define CM_SPK4_SHIFT 5 +#define CM_REAR2LIN 0x20 /* lin-in -> rear line out */ +#define CM_REAR2LIN_SHIFT 5 #define CM_REAR2FRONT 0x10 /* exchange rear/front */ #define CM_REAR2FRONT_SHIFT 4 #define CM_WAVEINL 0x08 /* digital wave rec. left chan */ @@ -276,12 +321,13 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address."); #define CM_VAUXR_MASK 0x0f #define CM_REG_MISC 0x27 +#define CM_UNKNOWN_27_MASK 0xd8 /* ? */ #define CM_XGPO1 0x20 // #define CM_XGPBIO 0x04 #define CM_MIC_CENTER_LFE 0x04 /* mic as center/lfe out? (model 039 or later?) */ #define CM_SPDIF_INVERSE 0x04 /* spdif input phase inverse (model 037) */ #define CM_SPDVALID 0x02 /* spdif input valid check */ -#define CM_DMAUTO 0x01 +#define CM_DMAUTO 0x01 /* SB16 DMA auto detect */ #define CM_REG_AC97 0x28 /* hmmm.. do we have ac97 link? */ /* @@ -322,18 +368,20 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address."); /* * extended registers */ -#define CM_REG_CH0_FRAME1 0x80 /* base address */ -#define CM_REG_CH0_FRAME2 0x84 +#define CM_REG_CH0_FRAME1 0x80 /* write: base address */ +#define CM_REG_CH0_FRAME2 0x84 /* read: current address */ #define CM_REG_CH1_FRAME1 0x88 /* 0-15: count of samples at bus master; buffer size */ #define CM_REG_CH1_FRAME2 0x8C /* 16-31: count of samples at codec; fragment size */ + #define CM_REG_EXT_MISC 0x90 -#define CM_REG_MISC_CTRL_8768 0x92 /* reg. name the same as 0x18 */ -#define CM_CHB3D8C 0x20 /* 7.1 channels support */ -#define CM_SPD32FMT 0x10 /* SPDIF/IN 32k */ -#define CM_ADC2SPDIF 0x08 /* ADC output to SPDIF/OUT */ -#define CM_SHAREADC 0x04 /* DAC in ADC as Center/LFE */ -#define CM_REALTCMP 0x02 /* monitor the CMPL/CMPR of ADC */ -#define CM_INVLRCK 0x01 /* invert ZVPORT's LRCK */ +#define CM_ADC48K44K 0x10000000 /* ADC parameters group, 0: 44k, 1: 48k */ +#define CM_CHB3D8C 0x00200000 /* 7.1 channels support */ +#define CM_SPD32FMT 0x00100000 /* SPDIF/IN 32k sample rate */ +#define CM_ADC2SPDIF 0x00080000 /* ADC output to SPDIF/OUT */ +#define CM_SHAREADC 0x00040000 /* DAC in ADC as Center/LFE */ +#define CM_REALTCMP 0x00020000 /* monitor the CMPL/CMPR of ADC */ +#define CM_INVLRCK 0x00010000 /* invert ZVPORT's LRCK */ +#define CM_UNKNOWN_90_MASK 0x0000FFFF /* ? */ /* * size of i/o region @@ -383,15 +431,14 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address."); struct cmipci_pcm { struct snd_pcm_substream *substream; - int running; /* dac/adc running? */ + u8 running; /* dac/adc running? */ + u8 fmt; /* format bits */ + u8 is_dac; + u8 needs_silencing; unsigned int dma_size; /* in frames */ - unsigned int period_size; /* in frames */ + unsigned int shift; + unsigned int ch; /* channel (0/1) */ unsigned int offset; /* physical address of the buffer */ - unsigned int fmt; /* format bits */ - int ch; /* channel (0/1) */ - unsigned int is_dac; /* is dac? */ - int bytes_per_frame; - int shift; }; /* mixer elements toggled/resumed during ac3 playback */ @@ -424,7 +471,6 @@ struct cmipci { int chip_version; int max_channels; - unsigned int has_dual_dac: 1; unsigned int can_ac3_sw: 1; unsigned int can_ac3_hw: 1; unsigned int can_multi_ch: 1; @@ -557,6 +603,9 @@ static unsigned int rates[] = { 5512, 11025, 22050, 44100, 8000, 16000, 32000, 4 static unsigned int snd_cmipci_rate_freq(unsigned int rate) { unsigned int i; + + if (rate > 48000) + rate /= 2; for (i = 0; i < ARRAY_SIZE(rates); i++) { if (rates[i] == rate) return i; @@ -671,19 +720,19 @@ static int snd_cmipci_hw_free(struct snd_pcm_substream *substream) /* */ -static unsigned int hw_channels[] = {1, 2, 4, 5, 6, 8}; +static unsigned int hw_channels[] = {1, 2, 4, 6, 8}; static struct snd_pcm_hw_constraint_list hw_constraints_channels_4 = { .count = 3, .list = hw_channels, .mask = 0, }; static struct snd_pcm_hw_constraint_list hw_constraints_channels_6 = { - .count = 5, + .count = 4, .list = hw_channels, .mask = 0, }; static struct snd_pcm_hw_constraint_list hw_constraints_channels_8 = { - .count = 6, + .count = 5, .list = hw_channels, .mask = 0, }; @@ -691,48 +740,37 @@ static struct snd_pcm_hw_constraint_list hw_constraints_channels_8 = { static int set_dac_channels(struct cmipci *cm, struct cmipci_pcm *rec, int channels) { if (channels > 2) { - if (! cm->can_multi_ch) + if (!cm->can_multi_ch || !rec->ch) return -EINVAL; if (rec->fmt != 0x03) /* stereo 16bit only */ return -EINVAL; + } + if (cm->can_multi_ch) { spin_lock_irq(&cm->reg_lock); - snd_cmipci_set_bit(cm, CM_REG_LEGACY_CTRL, CM_NXCHG); - snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_XCHGDAC); - if (channels > 4) { - snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_CHB3D); - snd_cmipci_set_bit(cm, CM_REG_CHFORMAT, CM_CHB3D5C); + if (channels > 2) { + snd_cmipci_set_bit(cm, CM_REG_LEGACY_CTRL, CM_NXCHG); + snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_XCHGDAC); } else { - snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_CHB3D5C); - snd_cmipci_set_bit(cm, CM_REG_CHFORMAT, CM_CHB3D); + snd_cmipci_clear_bit(cm, CM_REG_LEGACY_CTRL, CM_NXCHG); + snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_XCHGDAC); } - if (channels >= 6) { + if (channels == 8) + snd_cmipci_set_bit(cm, CM_REG_EXT_MISC, CM_CHB3D8C); + else + snd_cmipci_clear_bit(cm, CM_REG_EXT_MISC, CM_CHB3D8C); + if (channels == 6) { + snd_cmipci_set_bit(cm, CM_REG_CHFORMAT, CM_CHB3D5C); snd_cmipci_set_bit(cm, CM_REG_LEGACY_CTRL, CM_CHB3D6C); - snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_ENCENTER); } else { - snd_cmipci_clear_bit(cm, CM_REG_LEGACY_CTRL, CM_CHB3D6C); - snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_ENCENTER); - } - if (cm->chip_version == 68) { - if (channels == 8) { - snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL_8768, CM_CHB3D8C); - } else { - snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL_8768, CM_CHB3D8C); - } - } - spin_unlock_irq(&cm->reg_lock); - - } else { - if (cm->can_multi_ch) { - spin_lock_irq(&cm->reg_lock); - snd_cmipci_clear_bit(cm, CM_REG_LEGACY_CTRL, CM_NXCHG); - snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_CHB3D); snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_CHB3D5C); snd_cmipci_clear_bit(cm, CM_REG_LEGACY_CTRL, CM_CHB3D6C); - snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_ENCENTER); - snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_XCHGDAC); - spin_unlock_irq(&cm->reg_lock); } + if (channels == 4) + snd_cmipci_set_bit(cm, CM_REG_CHFORMAT, CM_CHB3D); + else + snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_CHB3D); + spin_unlock_irq(&cm->reg_lock); } return 0; } @@ -746,6 +784,7 @@ static int snd_cmipci_pcm_prepare(struct cmipci *cm, struct cmipci_pcm *rec, struct snd_pcm_substream *substream) { unsigned int reg, freq, val; + unsigned int period_size; struct snd_pcm_runtime *runtime = substream->runtime; rec->fmt = 0; @@ -765,11 +804,11 @@ static int snd_cmipci_pcm_prepare(struct cmipci *cm, struct cmipci_pcm *rec, rec->offset = runtime->dma_addr; /* buffer and period sizes in frame */ rec->dma_size = runtime->buffer_size << rec->shift; - rec->period_size = runtime->period_size << rec->shift; + period_size = runtime->period_size << rec->shift; if (runtime->channels > 2) { /* multi-channels */ rec->dma_size = (rec->dma_size * runtime->channels) / 2; - rec->period_size = (rec->period_size * runtime->channels) / 2; + period_size = (period_size * runtime->channels) / 2; } spin_lock_irq(&cm->reg_lock); @@ -780,7 +819,7 @@ static int snd_cmipci_pcm_prepare(struct cmipci *cm, struct cmipci_pcm *rec, /* program sample counts */ reg = rec->ch ? CM_REG_CH1_FRAME2 : CM_REG_CH0_FRAME2; snd_cmipci_write_w(cm, reg, rec->dma_size - 1); - snd_cmipci_write_w(cm, reg + 2, rec->period_size - 1); + snd_cmipci_write_w(cm, reg + 2, period_size - 1); /* set adc/dac flag */ val = rec->ch ? CM_CHADC1 : CM_CHADC0; @@ -795,11 +834,11 @@ static int snd_cmipci_pcm_prepare(struct cmipci *cm, struct cmipci_pcm *rec, freq = snd_cmipci_rate_freq(runtime->rate); val = snd_cmipci_read(cm, CM_REG_FUNCTRL1); if (rec->ch) { - val &= ~CM_ASFC_MASK; - val |= (freq << CM_ASFC_SHIFT) & CM_ASFC_MASK; - } else { val &= ~CM_DSFC_MASK; val |= (freq << CM_DSFC_SHIFT) & CM_DSFC_MASK; + } else { + val &= ~CM_ASFC_MASK; + val |= (freq << CM_ASFC_SHIFT) & CM_ASFC_MASK; } snd_cmipci_write(cm, CM_REG_FUNCTRL1, val); //snd_printd("cmipci: functrl1 = %08x\n", val); @@ -813,6 +852,16 @@ static int snd_cmipci_pcm_prepare(struct cmipci *cm, struct cmipci_pcm *rec, val &= ~CM_CH0FMT_MASK; val |= rec->fmt << CM_CH0FMT_SHIFT; } + if (cm->chip_version == 68) { + if (runtime->rate == 88200) + val |= CM_CH0_SRATE_88K << (rec->ch * 2); + else + val &= ~(CM_CH0_SRATE_88K << (rec->ch * 2)); + if (runtime->rate == 96000) + val |= CM_CH0_SRATE_96K << (rec->ch * 2); + else + val &= ~(CM_CH0_SRATE_96K << (rec->ch * 2)); + } snd_cmipci_write(cm, CM_REG_CHFORMAT, val); //snd_printd("cmipci: chformat = %08x\n", val); @@ -826,7 +875,7 @@ static int snd_cmipci_pcm_prepare(struct cmipci *cm, struct cmipci_pcm *rec, * PCM trigger/stop */ static int snd_cmipci_pcm_trigger(struct cmipci *cm, struct cmipci_pcm *rec, - struct snd_pcm_substream *substream, int cmd) + int cmd) { unsigned int inthld, chen, reset, pause; int result = 0; @@ -855,6 +904,7 @@ static int snd_cmipci_pcm_trigger(struct cmipci *cm, struct cmipci_pcm *rec, cm->ctrl &= ~chen; snd_cmipci_write(cm, CM_REG_FUNCTRL0, cm->ctrl | reset); snd_cmipci_write(cm, CM_REG_FUNCTRL0, cm->ctrl & ~reset); + rec->needs_silencing = rec->is_dac; break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: @@ -906,7 +956,7 @@ static int snd_cmipci_playback_trigger(struct snd_pcm_substream *substream, int cmd) { struct cmipci *cm = snd_pcm_substream_chip(substream); - return snd_cmipci_pcm_trigger(cm, &cm->channel[CM_CH_PLAY], substream, cmd); + return snd_cmipci_pcm_trigger(cm, &cm->channel[CM_CH_PLAY], cmd); } static snd_pcm_uframes_t snd_cmipci_playback_pointer(struct snd_pcm_substream *substream) @@ -925,7 +975,7 @@ static int snd_cmipci_capture_trigger(struct snd_pcm_substream *substream, int cmd) { struct cmipci *cm = snd_pcm_substream_chip(substream); - return snd_cmipci_pcm_trigger(cm, &cm->channel[CM_CH_CAPT], substream, cmd); + return snd_cmipci_pcm_trigger(cm, &cm->channel[CM_CH_CAPT], cmd); } static snd_pcm_uframes_t snd_cmipci_capture_pointer(struct snd_pcm_substream *substream) @@ -1199,15 +1249,19 @@ static int setup_spdif_playback(struct cmipci *cm, struct snd_pcm_substream *sub snd_cmipci_set_bit(cm, CM_REG_FUNCTRL1, CM_PLAYBACK_SPDF); setup_ac3(cm, subs, do_ac3, rate); - if (rate == 48000) + if (rate == 48000 || rate == 96000) snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_SPDIF48K | CM_SPDF_AC97); else snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_SPDIF48K | CM_SPDF_AC97); - + if (rate > 48000) + snd_cmipci_set_bit(cm, CM_REG_CHFORMAT, CM_DBLSPDS); + else + snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_DBLSPDS); } else { /* they are controlled via "IEC958 Output Switch" */ /* snd_cmipci_clear_bit(cm, CM_REG_LEGACY_CTRL, CM_ENSPDOUT); */ /* snd_cmipci_clear_bit(cm, CM_REG_FUNCTRL1, CM_SPDO2DAC); */ + snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_DBLSPDS); snd_cmipci_clear_bit(cm, CM_REG_FUNCTRL1, CM_PLAYBACK_SPDF); setup_ac3(cm, subs, 0, 0); } @@ -1227,7 +1281,7 @@ static int snd_cmipci_playback_prepare(struct snd_pcm_substream *substream) int rate = substream->runtime->rate; int err, do_spdif, do_ac3 = 0; - do_spdif = ((rate == 44100 || rate == 48000) && + do_spdif = (rate >= 44100 && substream->runtime->format == SNDRV_PCM_FORMAT_S16_LE && substream->runtime->channels == 2); if (do_spdif && cm->can_ac3_hw) @@ -1252,11 +1306,75 @@ static int snd_cmipci_playback_spdif_prepare(struct snd_pcm_substream *substream return snd_cmipci_pcm_prepare(cm, &cm->channel[CM_CH_PLAY], substream); } +/* + * Apparently, the samples last played on channel A stay in some buffer, even + * after the channel is reset, and get added to the data for the rear DACs when + * playing a multichannel stream on channel B. This is likely to generate + * wraparounds and thus distortions. + * To avoid this, we play at least one zero sample after the actual stream has + * stopped. + */ +static void snd_cmipci_silence_hack(struct cmipci *cm, struct cmipci_pcm *rec) +{ + struct snd_pcm_runtime *runtime = rec->substream->runtime; + unsigned int reg, val; + + if (rec->needs_silencing && runtime && runtime->dma_area) { + /* set up a small silence buffer */ + memset(runtime->dma_area, 0, PAGE_SIZE); + reg = rec->ch ? CM_REG_CH1_FRAME2 : CM_REG_CH0_FRAME2; + val = ((PAGE_SIZE / 4) - 1) | (((PAGE_SIZE / 4) / 2 - 1) << 16); + snd_cmipci_write(cm, reg, val); + + /* configure for 16 bits, 2 channels, 8 kHz */ + if (runtime->channels > 2) + set_dac_channels(cm, rec, 2); + spin_lock_irq(&cm->reg_lock); + val = snd_cmipci_read(cm, CM_REG_FUNCTRL1); + val &= ~(CM_ASFC_MASK << (rec->ch * 3)); + val |= (4 << CM_ASFC_SHIFT) << (rec->ch * 3); + snd_cmipci_write(cm, CM_REG_FUNCTRL1, val); + val = snd_cmipci_read(cm, CM_REG_CHFORMAT); + val &= ~(CM_CH0FMT_MASK << (rec->ch * 2)); + val |= (3 << CM_CH0FMT_SHIFT) << (rec->ch * 2); + if (cm->chip_version == 68) { + val &= ~(CM_CH0_SRATE_88K << (rec->ch * 2)); + val &= ~(CM_CH0_SRATE_96K << (rec->ch * 2)); + } + snd_cmipci_write(cm, CM_REG_CHFORMAT, val); + + /* start stream (we don't need interrupts) */ + cm->ctrl |= CM_CHEN0 << rec->ch; + snd_cmipci_write(cm, CM_REG_FUNCTRL0, cm->ctrl); + spin_unlock_irq(&cm->reg_lock); + + msleep(1); + + /* stop and reset stream */ + spin_lock_irq(&cm->reg_lock); + cm->ctrl &= ~(CM_CHEN0 << rec->ch); + val = CM_RST_CH0 << rec->ch; + snd_cmipci_write(cm, CM_REG_FUNCTRL0, cm->ctrl | val); + snd_cmipci_write(cm, CM_REG_FUNCTRL0, cm->ctrl & ~val); + spin_unlock_irq(&cm->reg_lock); + + rec->needs_silencing = 0; + } +} + static int snd_cmipci_playback_hw_free(struct snd_pcm_substream *substream) { struct cmipci *cm = snd_pcm_substream_chip(substream); setup_spdif_playback(cm, substream, 0, 0); restore_mixer_state(cm); + snd_cmipci_silence_hack(cm, &cm->channel[0]); + return snd_cmipci_hw_free(substream); +} + +static int snd_cmipci_playback2_hw_free(struct snd_pcm_substream *substream) +{ + struct cmipci *cm = snd_pcm_substream_chip(substream); + snd_cmipci_silence_hack(cm, &cm->channel[1]); return snd_cmipci_hw_free(substream); } @@ -1515,7 +1633,11 @@ static int snd_cmipci_playback_open(struct snd_pcm_substream *substream) if ((err = open_device_check(cm, CM_OPEN_PLAYBACK, substream)) < 0) return err; runtime->hw = snd_cmipci_playback; - runtime->hw.channels_max = cm->max_channels; + if (cm->chip_version == 68) { + runtime->hw.rates |= SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000; + runtime->hw.rate_max = 96000; + } snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 0, 0x10000); cm->dig_pcm_status = cm->dig_status; return 0; @@ -1558,9 +1680,14 @@ static int snd_cmipci_playback2_open(struct snd_pcm_substream *substream) else if (cm->max_channels == 8) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &hw_constraints_channels_8); } - snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 0, 0x10000); } mutex_unlock(&cm->open_mutex); + if (cm->chip_version == 68) { + runtime->hw.rates |= SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000; + runtime->hw.rate_max = 96000; + } + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 0, 0x10000); return 0; } @@ -1574,8 +1701,15 @@ static int snd_cmipci_playback_spdif_open(struct snd_pcm_substream *substream) return err; if (cm->can_ac3_hw) { runtime->hw = snd_cmipci_playback_spdif; - if (cm->chip_version >= 37) + if (cm->chip_version >= 37) { runtime->hw.formats |= SNDRV_PCM_FMTBIT_S32_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); + } + if (cm->chip_version == 68) { + runtime->hw.rates |= SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000; + runtime->hw.rate_max = 96000; + } } else { runtime->hw = snd_cmipci_playback_iec958_subframe; } @@ -1668,7 +1802,7 @@ static struct snd_pcm_ops snd_cmipci_playback2_ops = { .close = snd_cmipci_playback2_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_cmipci_playback2_hw_params, - .hw_free = snd_cmipci_hw_free, + .hw_free = snd_cmipci_playback2_hw_free, .prepare = snd_cmipci_capture_prepare, /* channel B */ .trigger = snd_cmipci_capture_trigger, /* channel B */ .pointer = snd_cmipci_capture_pointer, /* channel B */ @@ -2139,15 +2273,7 @@ struct cmipci_switch_args { */ }; -static int snd_cmipci_uswitch_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_cmipci_uswitch_info snd_ctl_boolean_mono_info static int _snd_cmipci_uswitch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol, @@ -2260,8 +2386,8 @@ DEFINE_SWITCH_ARG(exchange_dac, CM_REG_MISC_CTRL, CM_XCHGDAC, 0, 0, 0); /* rever DEFINE_SWITCH_ARG(exchange_dac, CM_REG_MISC_CTRL, CM_XCHGDAC, CM_XCHGDAC, 0, 0); #endif DEFINE_BIT_SWITCH_ARG(fourch, CM_REG_MISC_CTRL, CM_N4SPK3D, 0, 0); -// DEFINE_BIT_SWITCH_ARG(line_rear, CM_REG_MIXER1, CM_SPK4, 1, 0); -// DEFINE_BIT_SWITCH_ARG(line_bass, CM_REG_LEGACY_CTRL, CM_LINE_AS_BASS, 0, 0); +// DEFINE_BIT_SWITCH_ARG(line_rear, CM_REG_MIXER1, CM_REAR2LIN, 1, 0); +// DEFINE_BIT_SWITCH_ARG(line_bass, CM_REG_LEGACY_CTRL, CM_CENTR2LIN|CM_BASE2LIN, 0, 0); // DEFINE_BIT_SWITCH_ARG(joystick, CM_REG_FUNCTRL1, CM_JYSTK_EN, 0, 0); /* now module option */ DEFINE_SWITCH_ARG(modem, CM_REG_MISC_CTRL, CM_FLINKON|CM_FLINKOFF, CM_FLINKON, 0, 0); @@ -2331,11 +2457,11 @@ static inline unsigned int get_line_in_mode(struct cmipci *cm) unsigned int val; if (cm->chip_version >= 39) { val = snd_cmipci_read(cm, CM_REG_LEGACY_CTRL); - if (val & CM_LINE_AS_BASS) + if (val & (CM_CENTR2LIN | CM_BASE2LIN)) return 2; } val = snd_cmipci_read_b(cm, CM_REG_MIXER1); - if (val & CM_SPK4) + if (val & CM_REAR2LIN) return 1; return 0; } @@ -2359,13 +2485,13 @@ static int snd_cmipci_line_in_mode_put(struct snd_kcontrol *kcontrol, spin_lock_irq(&cm->reg_lock); if (ucontrol->value.enumerated.item[0] == 2) - change = snd_cmipci_set_bit(cm, CM_REG_LEGACY_CTRL, CM_LINE_AS_BASS); + change = snd_cmipci_set_bit(cm, CM_REG_LEGACY_CTRL, CM_CENTR2LIN | CM_BASE2LIN); else - change = snd_cmipci_clear_bit(cm, CM_REG_LEGACY_CTRL, CM_LINE_AS_BASS); + change = snd_cmipci_clear_bit(cm, CM_REG_LEGACY_CTRL, CM_CENTR2LIN | CM_BASE2LIN); if (ucontrol->value.enumerated.item[0] == 1) - change |= snd_cmipci_set_bit_b(cm, CM_REG_MIXER1, CM_SPK4); + change |= snd_cmipci_set_bit_b(cm, CM_REG_MIXER1, CM_REAR2LIN); else - change |= snd_cmipci_clear_bit_b(cm, CM_REG_MIXER1, CM_SPK4); + change |= snd_cmipci_clear_bit_b(cm, CM_REG_MIXER1, CM_REAR2LIN); spin_unlock_irq(&cm->reg_lock); return change; } @@ -2583,19 +2709,18 @@ static void snd_cmipci_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct cmipci *cm = entry->private_data; - int i; + int i, v; - snd_iprintf(buffer, "%s\n\n", cm->card->longname); - for (i = 0; i < 0x40; i++) { - int v = inb(cm->iobase + i); + snd_iprintf(buffer, "%s\n", cm->card->longname); + for (i = 0; i < 0x94; i++) { + if (i == 0x28) + i = 0x90; + v = inb(cm->iobase + i); if (i % 4 == 0) - snd_iprintf(buffer, "%02x: ", i); - snd_iprintf(buffer, "%02x", v); - if (i % 4 == 3) - snd_iprintf(buffer, "\n"); - else - snd_iprintf(buffer, " "); + snd_iprintf(buffer, "\n%02x:", i); + snd_iprintf(buffer, " %02x", v); } + snd_iprintf(buffer, "\n"); } static void __devinit snd_cmipci_proc_init(struct cmipci *cm) @@ -2633,46 +2758,40 @@ static void __devinit query_chip(struct cmipci *cm) if (! detect) { /* check reg 08h, bit 24-28 */ detect = snd_cmipci_read(cm, CM_REG_CHFORMAT) & CM_CHIP_MASK1; - if (! detect) { + switch (detect) { + case 0: cm->chip_version = 33; - cm->max_channels = 2; if (cm->do_soft_ac3) cm->can_ac3_sw = 1; else cm->can_ac3_hw = 1; - cm->has_dual_dac = 1; - } else { + break; + case CM_CHIP_037: cm->chip_version = 37; - cm->max_channels = 2; cm->can_ac3_hw = 1; - cm->has_dual_dac = 1; + break; + default: + cm->chip_version = 39; + cm->can_ac3_hw = 1; + break; } + cm->max_channels = 2; } else { - /* check reg 0Ch, bit 26 */ - if (detect & CM_CHIP_8768) { - cm->chip_version = 68; - cm->max_channels = 8; - cm->can_ac3_hw = 1; - cm->has_dual_dac = 1; - cm->can_multi_ch = 1; - } else if (detect & CM_CHIP_055) { - cm->chip_version = 55; - cm->max_channels = 6; - cm->can_ac3_hw = 1; - cm->has_dual_dac = 1; - cm->can_multi_ch = 1; - } else if (detect & CM_CHIP_039) { + if (detect & CM_CHIP_039) { cm->chip_version = 39; if (detect & CM_CHIP_039_6CH) /* 4 or 6 channels */ cm->max_channels = 6; else cm->max_channels = 4; - cm->can_ac3_hw = 1; - cm->has_dual_dac = 1; - cm->can_multi_ch = 1; + } else if (detect & CM_CHIP_8768) { + cm->chip_version = 68; + cm->max_channels = 8; } else { - printk(KERN_ERR "chip %x version not supported\n", detect); + cm->chip_version = 55; + cm->max_channels = 6; } + cm->can_ac3_hw = 1; + cm->can_multi_ch = 1; } } @@ -2782,10 +2901,14 @@ static int __devinit snd_cmipci_create_fm(struct cmipci *cm, long fm_port) if (!fm_port) goto disable_fm; - /* first try FM regs in PCI port range */ - iosynth = cm->iobase + CM_REG_FM_PCI; - err = snd_opl3_create(cm->card, iosynth, iosynth + 2, - OPL3_HW_OPL3, 1, &opl3); + if (cm->chip_version >= 39) { + /* first try FM regs in PCI port range */ + iosynth = cm->iobase + CM_REG_FM_PCI; + err = snd_opl3_create(cm->card, iosynth, iosynth + 2, + OPL3_HW_OPL3, 1, &opl3); + } else { + err = -EIO; + } if (err < 0) { /* then try legacy ports */ val = snd_cmipci_read(cm, CM_REG_LEGACY_CTRL) & ~CM_FMSEL_MASK; @@ -2829,9 +2952,10 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc static struct snd_device_ops ops = { .dev_free = snd_cmipci_dev_free, }; - unsigned int val = 0; + unsigned int val; long iomidi; - int integrated_midi; + int integrated_midi = 0; + char modelstr[16]; int pcm_index, pcm_spdif_index; static struct pci_device_id intel_82437vx[] = { { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_82437VX) }, @@ -2904,6 +3028,8 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc #endif /* initialize codec registers */ + snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_RESET); + snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_RESET); snd_cmipci_write(cm, CM_REG_INT_HLDCLR, 0); /* disable ints */ snd_cmipci_ch_reset(cm, CM_CH_PLAY); snd_cmipci_ch_reset(cm, CM_CH_CAPT); @@ -2917,6 +3043,10 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc #else snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_XCHGDAC); #endif + if (cm->chip_version) { + snd_cmipci_write_b(cm, CM_REG_EXT_MISC, 0x20); /* magic */ + snd_cmipci_write_b(cm, CM_REG_EXT_MISC + 1, 0x09); /* more magic */ + } /* Set Bus Master Request */ snd_cmipci_set_bit(cm, CM_REG_FUNCTRL1, CM_BREQ); @@ -2931,15 +3061,55 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc break; } + if (cm->chip_version < 68) { + val = pci->device < 0x110 ? 8338 : 8738; + } else { + switch (snd_cmipci_read_b(cm, CM_REG_INT_HLDCLR + 3) & 0x03) { + case 0: + val = 8769; + break; + case 2: + val = 8762; + break; + default: + switch ((pci->subsystem_vendor << 16) | + pci->subsystem_device) { + case 0x13f69761: + case 0x584d3741: + case 0x584d3751: + case 0x584d3761: + case 0x584d3771: + case 0x72848384: + val = 8770; + break; + default: + val = 8768; + break; + } + } + } + sprintf(card->shortname, "C-Media CMI%d", val); + if (cm->chip_version < 68) + sprintf(modelstr, " (model %d)", cm->chip_version); + else + modelstr[0] = '\0'; + sprintf(card->longname, "%s%s at %#lx, irq %i", + card->shortname, modelstr, cm->iobase, cm->irq); + if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, cm, &ops)) < 0) { snd_cmipci_free(cm); return err; } - integrated_midi = snd_cmipci_read_b(cm, CM_REG_MPU_PCI) != 0xff; - if (integrated_midi && mpu_port[dev] == 1) - iomidi = cm->iobase + CM_REG_MPU_PCI; - else { + if (cm->chip_version >= 39) { + val = snd_cmipci_read_b(cm, CM_REG_MPU_PCI + 1); + if (val != 0x00 && val != 0xff) { + iomidi = cm->iobase + CM_REG_MPU_PCI; + integrated_midi = 1; + } + } + if (!integrated_midi) { + val = 0; iomidi = mpu_port[dev]; switch (iomidi) { case 0x320: val = CM_VMPU_320; break; @@ -2953,11 +3123,21 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc snd_cmipci_write(cm, CM_REG_LEGACY_CTRL, val); /* enable UART */ snd_cmipci_set_bit(cm, CM_REG_FUNCTRL1, CM_UART_EN); + if (inb(iomidi + 1) == 0xff) { + snd_printk(KERN_ERR "cannot enable MPU-401 port" + " at %#lx\n", iomidi); + snd_cmipci_clear_bit(cm, CM_REG_FUNCTRL1, + CM_UART_EN); + iomidi = 0; + } } } - if ((err = snd_cmipci_create_fm(cm, fm_port[dev])) < 0) - return err; + if (cm->chip_version < 68) { + err = snd_cmipci_create_fm(cm, fm_port[dev]); + if (err < 0) + return err; + } /* reset mixer */ snd_cmipci_mixer_write(cm, 0, 0); @@ -2969,11 +3149,9 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc if ((err = snd_cmipci_pcm_new(cm, pcm_index)) < 0) return err; pcm_index++; - if (cm->has_dual_dac) { - if ((err = snd_cmipci_pcm2_new(cm, pcm_index)) < 0) - return err; - pcm_index++; - } + if ((err = snd_cmipci_pcm2_new(cm, pcm_index)) < 0) + return err; + pcm_index++; if (cm->can_ac3_hw || cm->can_ac3_sw) { pcm_spdif_index = pcm_index; if ((err = snd_cmipci_pcm_spdif_new(cm, pcm_index)) < 0) @@ -3057,15 +3235,6 @@ static int __devinit snd_cmipci_probe(struct pci_dev *pci, } card->private_data = cm; - sprintf(card->shortname, "C-Media PCI %s", card->driver); - sprintf(card->longname, "%s (model %d) at 0x%lx, irq %i", - card->shortname, - cm->chip_version, - cm->iobase, - cm->irq); - - //snd_printd("%s is detected\n", card->longname); - if ((err = snd_card_register(card)) < 0) { snd_card_free(card); return err; diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index 44cf54607647..9a55f4a9739b 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -1,6 +1,6 @@ /* * Driver for Cirrus Logic CS4281 based PCI soundcard - * Copyright (c) by Jaroslav Kysela <perex@suse.cz>, + * Copyright (c) by Jaroslav Kysela <perex@perex.cz>, * * * This program is free software; you can redistribute it and/or modify @@ -38,7 +38,7 @@ #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Cirrus Logic CS4281"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Cirrus Logic,CS4281}}"); @@ -842,12 +842,11 @@ static snd_pcm_uframes_t snd_cs4281_pointer(struct snd_pcm_substream *substream) static struct snd_pcm_hardware snd_cs4281_playback = { - .info = (SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME | - SNDRV_PCM_INFO_SYNC_START), + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_BE | SNDRV_PCM_FMTBIT_S16_BE | @@ -868,12 +867,11 @@ static struct snd_pcm_hardware snd_cs4281_playback = static struct snd_pcm_hardware snd_cs4281_capture = { - .info = (SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME | - SNDRV_PCM_INFO_SYNC_START), + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_BE | SNDRV_PCM_FMTBIT_S16_BE | @@ -904,7 +902,6 @@ static int snd_cs4281_playback_open(struct snd_pcm_substream *substream) dma->right_slot = 1; runtime->private_data = dma; runtime->hw = snd_cs4281_playback; - snd_pcm_set_sync(substream); /* should be detected from the AC'97 layer, but it seems that although CS4297A rev B reports 18-bit ADC resolution, samples are 20-bit */ @@ -924,7 +921,6 @@ static int snd_cs4281_capture_open(struct snd_pcm_substream *substream) dma->right_slot = 11; runtime->private_data = dma; runtime->hw = snd_cs4281_capture; - snd_pcm_set_sync(substream); /* should be detected from the AC'97 layer, but it seems that although CS4297A rev B reports 18-bit ADC resolution, samples are 20-bit */ diff --git a/sound/pci/cs46xx/Makefile b/sound/pci/cs46xx/Makefile index d8b77b89aec4..67e811ec8539 100644 --- a/sound/pci/cs46xx/Makefile +++ b/sound/pci/cs46xx/Makefile @@ -1,12 +1,10 @@ # # Makefile for ALSA -# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz> +# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz> # -snd-cs46xx-objs := cs46xx.o cs46xx_lib.o -ifeq ($(CONFIG_SND_CS46XX_NEW_DSP),y) - snd-cs46xx-objs += dsp_spos.o dsp_spos_scb_lib.o -endif +snd-cs46xx-y := cs46xx.o cs46xx_lib.o +snd-cs46xx-$(CONFIG_SND_CS46XX_NEW_DSP) += dsp_spos.o dsp_spos_scb_lib.o # Toplevel Module Dependency obj-$(CONFIG_SND_CS46XX) += snd-cs46xx.o diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c index 8b6cd144d101..2699cb6c2cd6 100644 --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -1,6 +1,6 @@ /* * The driver for the Cirrus Logic's Sound Fusion CS46XX based soundcards - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify @@ -34,7 +34,7 @@ #include <sound/cs46xx.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Cirrus Logic Sound Fusion CS46XX"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Cirrus Logic,Sound Fusion (CS4280)}," diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 71d7aab9d869..2c7bfc9fef61 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Abramo Bagnara <abramo@alsa-project.org> * Cirrus Logic, Inc. * Routines for control of Cirrus Logic CS461x chips @@ -1818,15 +1818,7 @@ static int snd_cs46xx_vol_iec958_put(struct snd_kcontrol *kcontrol, struct snd_c } #endif -static int snd_mixer_boolean_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_mixer_boolean_info snd_ctl_boolean_mono_info static int snd_cs46xx_iec958_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff --git a/sound/pci/cs46xx/cs46xx_lib.h b/sound/pci/cs46xx/cs46xx_lib.h index 20dcd72f06c1..018a7de56017 100644 --- a/sound/pci/cs46xx/cs46xx_lib.h +++ b/sound/pci/cs46xx/cs46xx_lib.h @@ -1,6 +1,6 @@ /* * The driver for the Cirrus Logic's Sound Fusion CS46XX based soundcards - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/sound/pci/cs46xx/dsp_spos.h b/sound/pci/cs46xx/dsp_spos.h index 0d246bca4184..f9e169d33c03 100644 --- a/sound/pci/cs46xx/dsp_spos.h +++ b/sound/pci/cs46xx/dsp_spos.h @@ -1,6 +1,6 @@ /* * The driver for the Cirrus Logic's Sound Fusion CS46XX based soundcards - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * * This program is free software; you can redistribute it and/or modify diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index 57e357de1500..eded4dfeba12 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -1480,7 +1480,7 @@ void cs46xx_dsp_destroy_pcm_channel (struct snd_cs46xx * chip, if (!pcm_channel->src_scb->ref_count) { cs46xx_dsp_remove_scb(chip,pcm_channel->src_scb); - snd_assert (pcm_channel->src_slot >= 0 && pcm_channel->src_slot <= DSP_MAX_SRC_NR, + snd_assert (pcm_channel->src_slot >= 0 && pcm_channel->src_slot < DSP_MAX_SRC_NR, return ); ins->src_scb_slots[pcm_channel->src_slot] = 0; diff --git a/sound/pci/cs5535audio/Makefile b/sound/pci/cs5535audio/Makefile index ad947b4c04cc..bb3d57e6a3cb 100644 --- a/sound/pci/cs5535audio/Makefile +++ b/sound/pci/cs5535audio/Makefile @@ -2,11 +2,8 @@ # Makefile for cs5535audio # -snd-cs5535audio-objs := cs5535audio.o cs5535audio_pcm.o - -ifeq ($(CONFIG_PM),y) -snd-cs5535audio-objs += cs5535audio_pm.o -endif +snd-cs5535audio-y := cs5535audio.o cs5535audio_pcm.o +snd-cs5535audio-$(CONFIG_PM) += cs5535audio_pm.o # Toplevel Module Dependency obj-$(CONFIG_SND_CS5535AUDIO) += snd-cs5535audio.o diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index b8e75ef9c1e6..2b35889787be 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -206,7 +206,6 @@ static void process_bm1_irq(struct cs5535audio *cs5535au) static irqreturn_t snd_cs5535audio_interrupt(int irq, void *dev_id) { u16 acc_irq_stat; - u8 bm_stat; unsigned char count; struct cs5535audio *cs5535au = dev_id; @@ -217,7 +216,7 @@ static irqreturn_t snd_cs5535audio_interrupt(int irq, void *dev_id) if (!acc_irq_stat) return IRQ_NONE; - for (count = 0; count < 10; count++) { + for (count = 0; count < 4; count++) { if (acc_irq_stat & (1 << count)) { switch (count) { case IRQ_STS: @@ -232,26 +231,9 @@ static irqreturn_t snd_cs5535audio_interrupt(int irq, void *dev_id) case BM1_IRQ_STS: process_bm1_irq(cs5535au); break; - case BM2_IRQ_STS: - bm_stat = cs_readb(cs5535au, ACC_BM2_STATUS); - break; - case BM3_IRQ_STS: - bm_stat = cs_readb(cs5535au, ACC_BM3_STATUS); - break; - case BM4_IRQ_STS: - bm_stat = cs_readb(cs5535au, ACC_BM4_STATUS); - break; - case BM5_IRQ_STS: - bm_stat = cs_readb(cs5535au, ACC_BM5_STATUS); - break; - case BM6_IRQ_STS: - bm_stat = cs_readb(cs5535au, ACC_BM6_STATUS); - break; - case BM7_IRQ_STS: - bm_stat = cs_readb(cs5535au, ACC_BM7_STATUS); - break; default: - snd_printk(KERN_ERR "Unexpected irq src\n"); + snd_printk(KERN_ERR "Unexpected irq src: " + "0x%x\n", acc_irq_stat); break; } } diff --git a/sound/pci/cs5535audio/cs5535audio.h b/sound/pci/cs5535audio/cs5535audio.h index 4fd1f31a6cf9..66bae7664193 100644 --- a/sound/pci/cs5535audio/cs5535audio.h +++ b/sound/pci/cs5535audio/cs5535audio.h @@ -16,57 +16,28 @@ #define ACC_IRQ_STATUS 0x12 #define ACC_BM0_CMD 0x20 #define ACC_BM1_CMD 0x28 -#define ACC_BM2_CMD 0x30 -#define ACC_BM3_CMD 0x38 -#define ACC_BM4_CMD 0x40 -#define ACC_BM5_CMD 0x48 -#define ACC_BM6_CMD 0x50 -#define ACC_BM7_CMD 0x58 #define ACC_BM0_PRD 0x24 #define ACC_BM1_PRD 0x2C -#define ACC_BM2_PRD 0x34 -#define ACC_BM3_PRD 0x3C -#define ACC_BM4_PRD 0x44 -#define ACC_BM5_PRD 0x4C -#define ACC_BM6_PRD 0x54 -#define ACC_BM7_PRD 0x5C #define ACC_BM0_STATUS 0x21 #define ACC_BM1_STATUS 0x29 -#define ACC_BM2_STATUS 0x31 -#define ACC_BM3_STATUS 0x39 -#define ACC_BM4_STATUS 0x41 -#define ACC_BM5_STATUS 0x49 -#define ACC_BM6_STATUS 0x51 -#define ACC_BM7_STATUS 0x59 #define ACC_BM0_PNTR 0x60 #define ACC_BM1_PNTR 0x64 -#define ACC_BM2_PNTR 0x68 -#define ACC_BM3_PNTR 0x6C -#define ACC_BM4_PNTR 0x70 -#define ACC_BM5_PNTR 0x74 -#define ACC_BM6_PNTR 0x78 -#define ACC_BM7_PNTR 0x7C + /* acc_codec bar0 reg bits */ /* ACC_IRQ_STATUS */ #define IRQ_STS 0 #define WU_IRQ_STS 1 #define BM0_IRQ_STS 2 #define BM1_IRQ_STS 3 -#define BM2_IRQ_STS 4 -#define BM3_IRQ_STS 5 -#define BM4_IRQ_STS 6 -#define BM5_IRQ_STS 7 -#define BM6_IRQ_STS 8 -#define BM7_IRQ_STS 9 /* ACC_BMX_STATUS */ #define EOP (1<<0) #define BM_EOP_ERR (1<<1) /* ACC_BMX_CTL */ -#define BM_CTL_EN 0x00000001 -#define BM_CTL_PAUSE 0x00000011 -#define BM_CTL_DIS 0x00000000 -#define BM_CTL_BYTE_ORD_LE 0x00000000 -#define BM_CTL_BYTE_ORD_BE 0x00000100 +#define BM_CTL_EN 0x01 +#define BM_CTL_PAUSE 0x03 +#define BM_CTL_DIS 0x00 +#define BM_CTL_BYTE_ORD_LE 0x00 +#define BM_CTL_BYTE_ORD_BE 0x04 /* cs5535 specific ac97 codec register defines */ #define CMD_MASK 0xFF00FFFF #define CMD_NEW 0x00010000 @@ -106,7 +77,6 @@ struct cs5535audio_dma { struct snd_pcm_substream *substream; unsigned int buf_addr, buf_bytes; unsigned int period_bytes, periods; - int suspended; u32 saved_prd; }; diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c index 5450a9e8f133..21df0634af32 100644 --- a/sound/pci/cs5535audio/cs5535audio_pcm.c +++ b/sound/pci/cs5535audio/cs5535audio_pcm.c @@ -43,7 +43,6 @@ static struct snd_pcm_hardware snd_cs5535audio_playback = SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_SYNC_START | SNDRV_PCM_INFO_RESUME ), .formats = ( @@ -71,8 +70,7 @@ static struct snd_pcm_hardware snd_cs5535audio_capture = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_SYNC_START + SNDRV_PCM_INFO_MMAP_VALID ), .formats = ( SNDRV_PCM_FMTBIT_S16_LE @@ -102,7 +100,6 @@ static int snd_cs5535audio_playback_open(struct snd_pcm_substream *substream) runtime->hw = snd_cs5535audio_playback; cs5535au->playback_substream = substream; runtime->private_data = &(cs5535au->dmas[CS5535AUDIO_DMA_PLAYBACK]); - snd_pcm_set_sync(substream); if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) return err; @@ -164,6 +161,7 @@ static int cs5535audio_build_dma_packets(struct cs5535audio *cs5535au, jmpprd_addr = cpu_to_le32(lastdesc->addr + (sizeof(struct cs5535audio_dma_desc)*periods)); + dma->substream = substream; dma->period_bytes = period_bytes; dma->periods = periods; spin_lock_irq(&cs5535au->reg_lock); @@ -241,6 +239,7 @@ static void cs5535audio_clear_dma_packets(struct cs5535audio *cs5535au, { snd_dma_free_pages(&dma->desc_buf); dma->desc_buf.area = NULL; + dma->substream = NULL; } static int snd_cs5535audio_hw_params(struct snd_pcm_substream *substream, @@ -298,14 +297,12 @@ static int snd_cs5535audio_trigger(struct snd_pcm_substream *substream, int cmd) break; case SNDRV_PCM_TRIGGER_RESUME: dma->ops->enable_dma(cs5535au); - dma->suspended = 0; break; case SNDRV_PCM_TRIGGER_STOP: dma->ops->disable_dma(cs5535au); break; case SNDRV_PCM_TRIGGER_SUSPEND: dma->ops->disable_dma(cs5535au); - dma->suspended = 1; break; default: snd_printk(KERN_ERR "unhandled trigger\n"); @@ -348,7 +345,6 @@ static int snd_cs5535audio_capture_open(struct snd_pcm_substream *substream) runtime->hw = snd_cs5535audio_capture; cs5535au->capture_substream = substream; runtime->private_data = &(cs5535au->dmas[CS5535AUDIO_DMA_CAPTURE]); - snd_pcm_set_sync(substream); if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) return err; diff --git a/sound/pci/cs5535audio/cs5535audio_pm.c b/sound/pci/cs5535audio/cs5535audio_pm.c index 3e4d198a4502..838708f6d45e 100644 --- a/sound/pci/cs5535audio/cs5535audio_pm.c +++ b/sound/pci/cs5535audio/cs5535audio_pm.c @@ -64,18 +64,21 @@ int snd_cs5535audio_suspend(struct pci_dev *pci, pm_message_t state) int i; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + snd_pcm_suspend_all(cs5535au->pcm); + snd_ac97_suspend(cs5535au->ac97); for (i = 0; i < NUM_CS5535AUDIO_DMAS; i++) { struct cs5535audio_dma *dma = &cs5535au->dmas[i]; - if (dma && dma->substream && !dma->suspended) + if (dma && dma->substream) dma->saved_prd = dma->ops->read_prd(cs5535au); } - snd_pcm_suspend_all(cs5535au->pcm); - snd_ac97_suspend(cs5535au->ac97); /* save important regs, then disable aclink in hw */ snd_cs5535audio_stop_hardware(cs5535au); + if (pci_save_state(pci)) { + printk(KERN_ERR "cs5535audio: pci_save_state failed!\n"); + return -EIO; + } pci_disable_device(pci); - pci_save_state(pci); pci_set_power_state(pci, pci_choose_state(pci, state)); return 0; } @@ -89,7 +92,12 @@ int snd_cs5535audio_resume(struct pci_dev *pci) int i; pci_set_power_state(pci, PCI_D0); - pci_restore_state(pci); + if (pci_restore_state(pci) < 0) { + printk(KERN_ERR "cs5535audio: pci_restore_state failed, " + "disabling device\n"); + snd_card_disconnect(card); + return -EIO; + } if (pci_enable_device(pci) < 0) { printk(KERN_ERR "cs5535audio: pci_enable_device failed, " "disabling device\n"); @@ -112,17 +120,17 @@ int snd_cs5535audio_resume(struct pci_dev *pci) if (!timeout) snd_printk(KERN_ERR "Failure getting AC Link ready\n"); - /* we depend on ac97 to perform the codec power up */ - snd_ac97_resume(cs5535au->ac97); /* set up rate regs, dma. actual initiation is done in trig */ for (i = 0; i < NUM_CS5535AUDIO_DMAS; i++) { struct cs5535audio_dma *dma = &cs5535au->dmas[i]; - if (dma && dma->substream && dma->suspended) { + if (dma && dma->substream) { dma->substream->ops->prepare(dma->substream); dma->ops->setup_prd(cs5535au, dma->saved_prd); } } - + + /* we depend on ac97 to perform the codec power up */ + snd_ac97_resume(cs5535au->ac97); snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index f27b6a733b96..499ee1a5319d 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -1595,15 +1595,7 @@ static struct snd_kcontrol_new snd_echo_clock_source_switch __devinitdata = { #ifdef ECHOCARD_HAS_PHANTOM_POWER /******************* Phantom power switch *******************/ -static int snd_echo_phantom_power_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_echo_phantom_power_info snd_ctl_boolean_mono_info static int snd_echo_phantom_power_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1646,15 +1638,7 @@ static struct snd_kcontrol_new snd_echo_phantom_power_switch __devinitdata = { #ifdef ECHOCARD_HAS_DIGITAL_IN_AUTOMUTE /******************* Digital input automute switch *******************/ -static int snd_echo_automute_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_echo_automute_info snd_ctl_boolean_mono_info static int snd_echo_automute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1695,18 +1679,7 @@ static struct snd_kcontrol_new snd_echo_automute_switch __devinitdata = { /******************* VU-meters switch *******************/ -static int snd_echo_vumeters_switch_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct echoaudio *chip; - - chip = snd_kcontrol_chip(kcontrol); - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_echo_vumeters_switch_info snd_ctl_boolean_mono_info static int snd_echo_vumeters_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index 42afa837d9b4..e6c100770392 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -43,11 +43,11 @@ static int wait_handshake(struct echoaudio *chip) { int i; - /* Wait up to 10ms for the handshake from the DSP */ + /* Wait up to 20ms for the handshake from the DSP */ for (i = 0; i < HANDSHAKE_TIMEOUT; i++) { /* Look for the handshake value */ + barrier(); if (chip->comm_page->handshake) { - /*if (i) DE_ACT(("Handshake time: %d\n", i));*/ return 0; } udelay(1); diff --git a/sound/pci/echoaudio/echoaudio_dsp.h b/sound/pci/echoaudio/echoaudio_dsp.h index e55ee00991ac..e352f3ae292c 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.h +++ b/sound/pci/echoaudio/echoaudio_dsp.h @@ -642,18 +642,18 @@ struct comm_page { /* Base Length*/ u32 flags; /* See Appendix A below 0x004 4 */ u32 unused; /* Unused entry 0x008 4 */ u32 sample_rate; /* Card sample rate in Hz 0x00c 4 */ - volatile u32 handshake; /* DSP command handshake 0x010 4 */ + u32 handshake; /* DSP command handshake 0x010 4 */ u32 cmd_start; /* Chs. to start mask 0x014 4 */ u32 cmd_stop; /* Chs. to stop mask 0x018 4 */ u32 cmd_reset; /* Chs. to reset mask 0x01c 4 */ u16 audio_format[DSP_MAXPIPES]; /* Chs. audio format 0x020 32*2 */ struct sg_entry sglist_addr[DSP_MAXPIPES]; /* Chs. Physical sglist addrs 0x060 32*8 */ - volatile u32 position[DSP_MAXPIPES]; + u32 position[DSP_MAXPIPES]; /* Positions for ea. ch. 0x160 32*4 */ - volatile s8 vu_meter[DSP_MAXPIPES]; + s8 vu_meter[DSP_MAXPIPES]; /* VU meters 0x1e0 32*1 */ - volatile s8 peak_meter[DSP_MAXPIPES]; + s8 peak_meter[DSP_MAXPIPES]; /* Peak meters 0x200 32*1 */ s8 line_out_level[DSP_MAXAUDIOOUTPUTS]; /* Output gain 0x220 16*1 */ @@ -665,7 +665,7 @@ struct comm_page { /* Base Length*/ /* Gina/Darla play filters - obsolete 0x3c0 168*4 */ u32 rec_coeff[MAX_REC_TAPS]; /* Gina/Darla record filters - obsolete 0x660 192*4 */ - volatile u16 midi_input[MIDI_IN_BUFFER_SIZE]; + u16 midi_input[MIDI_IN_BUFFER_SIZE]; /* MIDI input data transfer buffer 0x960 256*2 */ u8 gd_clock_state; /* Chg Gina/Darla clock state 0xb60 1 */ u8 gd_spdif_status; /* Chg. Gina/Darla S/PDIF state 0xb61 1 */ @@ -674,11 +674,10 @@ struct comm_page { /* Base Length*/ u32 nominal_level_mask; /* -10 level enable mask 0xb64 4 */ u16 input_clock; /* Chg. Input clock state 0xb68 2 */ u16 output_clock; /* Chg. Output clock state 0xb6a 2 */ - volatile u32 status_clocks; - /* Current Input clock state 0xb6c 4 */ + u32 status_clocks; /* Current Input clock state 0xb6c 4 */ u32 ext_box_status; /* External box status 0xb70 4 */ u32 cmd_add_buffer; /* Pipes to add (obsolete) 0xb74 4 */ - volatile u32 midi_out_free_count; + u32 midi_out_free_count; /* # of bytes free in MIDI output FIFO 0xb78 4 */ u32 unused2; /* Cyclic pipes 0xb7c 4 */ u32 control_register; diff --git a/sound/pci/emu10k1/Makefile b/sound/pci/emu10k1/Makefile index e521c38cef45..cf2d5636d8be 100644 --- a/sound/pci/emu10k1/Makefile +++ b/sound/pci/emu10k1/Makefile @@ -1,6 +1,6 @@ # # Makefile for ALSA -# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz> +# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz> # snd-emu10k1-objs := emu10k1.o emu10k1_main.o \ diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 55caf341933a..9680caff90c8 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -1,6 +1,6 @@ /* * The driver for the EMU10K1 (SB Live!) based soundcards - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * Copyright (c) by James Courtier-Dutton <James@superbug.demon.co.uk> * Added support for Audigy 2 Value. @@ -32,7 +32,7 @@ #include <sound/emu10k1.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("EMU10K1"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Creative Labs,SB Live!/PCI512/E-mu APS}," diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 404ae1be0a4b..97c41d72a255 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Creative Labs, Inc. * Routines for control of EMU10K1 chips * @@ -31,6 +31,8 @@ * */ +#include <linux/sched.h> +#include <linux/kthread.h> #include <sound/driver.h> #include <linux/delay.h> #include <linux/init.h> @@ -702,6 +704,65 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * file return 0; } +int emu1010_firmware_thread(void *data) { + struct snd_emu10k1 * emu = data; + int tmp,tmp2; + int reg; + int err; + + for (;;) { + /* Delay to allow Audio Dock to settle */ + msleep(1000); + if (kthread_should_stop()) + break; + snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &tmp ); /* IRQ Status */ + snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ® ); /* OPTIONS: Which cards are attached to the EMU */ + if (reg & EMU_HANA_OPTION_DOCK_OFFLINE) { + /* Audio Dock attached */ + /* Return to Audio Dock programming mode */ + snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware\n"); + snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_AUDIODOCK ); + if (emu->card_capabilities->emu1010 == 1) { + if ((err = snd_emu1010_load_firmware(emu, DOCK_FILENAME)) != 0) { + return err; + } + } else if (emu->card_capabilities->emu1010 == 2) { + if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) { + return err; + } + } else if (emu->card_capabilities->emu1010 == 3) { + if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) { + return err; + } + } + + snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0 ); + snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, ® ); + snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_IRQ_STATUS=0x%x\n",reg); + /* ID, should read & 0x7f = 0x55 when FPGA programmed. */ + snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); + snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_ID=0x%x\n",reg); + if ((reg & 0x1f) != 0x15) { + /* FPGA failed to be programmed */ + snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware file failed, reg=0x%x\n", reg); + return 0; + return -ENODEV; + } + snd_printk(KERN_INFO "emu1010: Audio Dock Firmware loaded\n"); + snd_emu1010_fpga_read(emu, EMU_DOCK_MAJOR_REV, &tmp ); + snd_emu1010_fpga_read(emu, EMU_DOCK_MINOR_REV, &tmp2 ); + snd_printk("Audio Dock ver:%d.%d\n",tmp ,tmp2); + /* Sync clocking between 1010 and Dock */ + /* Allow DLL to settle */ + msleep(10); + /* Unmute all. Default is muted after a firmware load */ + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE ); + break; + } + } + return 0; +} + /* * EMU-1010 - details found out from this driver, official MS Win drivers, * testing the card: @@ -817,8 +878,16 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ® ); snd_printk(KERN_INFO "emu1010: Card options=0x%x\n",reg); snd_emu1010_fpga_read(emu, EMU_HANA_OPTICAL_TYPE, &tmp ); - /* ADAT input. */ - snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, 0x01 ); + /* Optical -> ADAT I/O */ + /* 0 : SPDIF + * 1 : ADAT + */ + emu->emu1010.optical_in = 1; /* IN_ADAT */ + emu->emu1010.optical_out = 1; /* IN_ADAT */ + tmp = 0; + tmp = (emu->emu1010.optical_in ? EMU_HANA_OPTICAL_IN_ADAT : 0) | + (emu->emu1010.optical_out ? EMU_HANA_OPTICAL_OUT_ADAT : 0); + snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, tmp ); snd_emu1010_fpga_read(emu, EMU_HANA_ADC_PADS, &tmp ); /* Set no attenuation on Audio Dock pads. */ snd_emu1010_fpga_write(emu, EMU_HANA_ADC_PADS, 0x00 ); @@ -1004,49 +1073,12 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) snd_emu1010_fpga_read(emu, EMU_HANA_SPDIF_MODE, &tmp ); snd_emu1010_fpga_write(emu, EMU_HANA_SPDIF_MODE, 0x10 ); /* SPDIF Format spdif (or 0x11 for aes/ebu) */ - /* Delay to allow Audio Dock to settle */ - msleep(100); - snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &tmp ); /* IRQ Status */ - snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ® ); /* OPTIONS: Which cards are attached to the EMU */ - /* FIXME: The loading of this should be able to happen any time, - * as the user can plug/unplug it at any time - */ - if (reg & (EMU_HANA_OPTION_DOCK_ONLINE | EMU_HANA_OPTION_DOCK_OFFLINE) ) { - /* Audio Dock attached */ - /* Return to Audio Dock programming mode */ - snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware\n"); - snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_AUDIODOCK ); - if (emu->card_capabilities->emu1010 == 1) { - if ((err = snd_emu1010_load_firmware(emu, DOCK_FILENAME)) != 0) { - return err; - } - } else if (emu->card_capabilities->emu1010 == 2) { - if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) { - return err; - } - } else if (emu->card_capabilities->emu1010 == 3) { - if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) { - return err; - } - } + /* Start Micro/Audio Dock firmware loader thread */ + emu->emu1010.firmware_thread = kthread_create(&emu1010_firmware_thread, + emu, + "emu1010_firmware"); + wake_up_process(emu->emu1010.firmware_thread); - snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0 ); - snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, ® ); - snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_IRQ_STATUS=0x%x\n",reg); - /* ID, should read & 0x7f = 0x55 when FPGA programmed. */ - snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); - snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_ID=0x%x\n",reg); - if ((reg & 0x3f) != 0x15) { - /* FPGA failed to be programmed */ - snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware file failed, reg=0x%x\n", reg); - return 0; - return -ENODEV; - } - snd_printk(KERN_INFO "emu1010: Audio Dock Firmware loaded\n"); - snd_emu1010_fpga_read(emu, EMU_DOCK_MAJOR_REV, &tmp ); - snd_emu1010_fpga_read(emu, EMU_DOCK_MINOR_REV, &tmp2 ); - snd_printk("Audio Dock ver:%d.%d\n",tmp ,tmp2); - } #if 0 snd_emu1010_fpga_link_dst_src_write(emu, EMU_DST_HAMOA_DAC_LEFT1, EMU_SRC_ALICE_EMU32B + 2); /* ALICE2 bus 0xa2 */ @@ -1132,7 +1164,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) emu->emu1010.output_source[23] = 28; /* TEMP: Select SPDIF in/out */ - snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, 0x0); /* Output spdif */ + //snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, 0x0); /* Output spdif */ /* TEMP: Select 48kHz SPDIF out */ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x0); /* Mute all */ @@ -1173,6 +1205,7 @@ static int snd_emu10k1_free(struct snd_emu10k1 *emu) if (emu->card_capabilities->emu1010) { /* Disable 48Volt power to Audio Dock */ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0 ); + kthread_stop(emu->emu1010.firmware_thread); } if (emu->memhdr) snd_util_memhdr_free(emu->memhdr); @@ -1722,8 +1755,9 @@ int __devinit snd_emu10k1_create(struct snd_card *card, goto error; } - emu->page_ptr_table = (void **)vmalloc(emu->max_cache_pages * sizeof(void*)); - emu->page_addr_table = (unsigned long*)vmalloc(emu->max_cache_pages * sizeof(unsigned long)); + emu->page_ptr_table = vmalloc(emu->max_cache_pages * sizeof(void *)); + emu->page_addr_table = vmalloc(emu->max_cache_pages * + sizeof(unsigned long)); if (emu->page_ptr_table == NULL || emu->page_addr_table == NULL) { err = -ENOMEM; goto error; diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index e4af7a9b808c..1ec7ebaff9e9 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1062,14 +1062,7 @@ static int __devinit snd_emu10k1x_proc_init(struct emu10k1x * emu) return 0; } -static int snd_emu10k1x_shared_spdif_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_emu10k1x_shared_spdif_info snd_ctl_boolean_mono_info static int snd_emu10k1x_shared_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 7206c0fa06f2..9bf1cd592199 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Creative Labs, Inc. * Routines for effect processor FX8010 * @@ -642,10 +642,8 @@ snd_emu10k1_look_for_ctl(struct snd_emu10k1 *emu, struct snd_ctl_elem_id *id) { struct snd_emu10k1_fx8010_ctl *ctl; struct snd_kcontrol *kcontrol; - struct list_head *list; - - list_for_each(list, &emu->fx8010.gpr_ctl) { - ctl = emu10k1_gpr_ctl(list); + + list_for_each_entry(ctl, &emu->fx8010.gpr_ctl, list) { kcontrol = ctl->kcontrol; if (kcontrol->id.iface == id->iface && !strcmp(kcontrol->id.name, id->name) && @@ -895,14 +893,12 @@ static int snd_emu10k1_list_controls(struct snd_emu10k1 *emu, struct snd_emu10k1_fx8010_control_gpr *gctl; struct snd_emu10k1_fx8010_ctl *ctl; struct snd_ctl_elem_id *id; - struct list_head *list; gctl = kmalloc(sizeof(*gctl), GFP_KERNEL); if (! gctl) return -ENOMEM; - list_for_each(list, &emu->fx8010.gpr_ctl) { - ctl = emu10k1_gpr_ctl(list); + list_for_each_entry(ctl, &emu->fx8010.gpr_ctl, list) { total++; if (icode->gpr_list_controls && i < icode->gpr_list_control_count) { @@ -1207,7 +1203,7 @@ static int __devinit _snd_emu10k1_audigy_init_efx(struct snd_emu10k1 *emu) A_OP(icode, &ptr, iMAC0, A_GPR(playback+1), A_C_00000000, A_GPR(gpr+1), A_FXBUS(FXBUS_PCM_RIGHT_FRONT)); snd_emu10k1_init_stereo_control(&controls[nctl++], "PCM Front Playback Volume", gpr, 100); gpr += 2; - + /* PCM Surround Playback (independent from stereo mix) */ A_OP(icode, &ptr, iMAC0, A_GPR(playback+2), A_C_00000000, A_GPR(gpr), A_FXBUS(FXBUS_PCM_LEFT_REAR)); A_OP(icode, &ptr, iMAC0, A_GPR(playback+3), A_C_00000000, A_GPR(gpr+1), A_FXBUS(FXBUS_PCM_RIGHT_REAR)); @@ -1267,8 +1263,16 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) /* emu1212 DSP 0 and DSP 1 Capture */ if (emu->card_capabilities->emu1010) { - A_OP(icode, &ptr, iMAC0, A_GPR(capture+0), A_GPR(capture+0), A_GPR(gpr), A_P16VIN(0x0)); - A_OP(icode, &ptr, iMAC0, A_GPR(capture+1), A_GPR(capture+1), A_GPR(gpr+1), A_P16VIN(0x1)); + if (emu->card_capabilities->ca0108_chip) { + /* Note:JCD:No longer bit shift lower 16bits to upper 16bits of 32bit value. */ + A_OP(icode, &ptr, iMACINT0, A_GPR(tmp), A_C_00000000, A3_EMU32IN(0x0), A_C_00000001); + A_OP(icode, &ptr, iMAC0, A_GPR(capture+0), A_GPR(capture+0), A_GPR(gpr), A_GPR(tmp)); + A_OP(icode, &ptr, iMACINT0, A_GPR(tmp), A_C_00000000, A3_EMU32IN(0x1), A_C_00000001); + A_OP(icode, &ptr, iMAC0, A_GPR(capture+1), A_GPR(capture+1), A_GPR(gpr), A_GPR(tmp)); + } else { + A_OP(icode, &ptr, iMAC0, A_GPR(capture+0), A_GPR(capture+0), A_GPR(gpr), A_P16VIN(0x0)); + A_OP(icode, &ptr, iMAC0, A_GPR(capture+1), A_GPR(capture+1), A_GPR(gpr+1), A_P16VIN(0x1)); + } snd_emu10k1_init_stereo_control(&controls[nctl++], "EMU Capture Volume", gpr, 0); gpr += 2; } @@ -1516,7 +1520,11 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) /* EMU1010 Outputs from PCM Front, Rear, Center, LFE, Side */ snd_printk("EMU outputs on\n"); for (z = 0; z < 8; z++) { - A_OP(icode, &ptr, iACC3, A_EMU32OUTL(z), A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + z), A_C_00000000, A_C_00000000); + if (emu->card_capabilities->ca0108_chip) { + A_OP(icode, &ptr, iACC3, A3_EMU32OUT(z), A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + z), A_C_00000000, A_C_00000000); + } else { + A_OP(icode, &ptr, iACC3, A_EMU32OUTL(z), A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + z), A_C_00000000, A_C_00000000); + } } } @@ -1557,106 +1565,116 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) #endif if (emu->card_capabilities->emu1010) { - snd_printk("EMU inputs on\n"); - /* Capture 16 (originally 8) channels of S32_LE sound */ - - /* printk("emufx.c: gpr=0x%x, tmp=0x%x\n",gpr, tmp); */ - /* For the EMU1010: How to get 32bit values from the DSP. High 16bits into L, low 16bits into R. */ - /* A_P16VIN(0) is delayed by one sample, - * so all other A_P16VIN channels will need to also be delayed - */ - /* Left ADC in. 1 of 2 */ - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_P16VIN(0x0), A_FXBUS2(0) ); - /* Right ADC in 1 of 2 */ - gpr_map[gpr++] = 0x00000000; - /* Delaying by one sample: instead of copying the input - * value A_P16VIN to output A_FXBUS2 as in the first channel, - * we use an auxiliary register, delaying the value by one - * sample - */ - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(2) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x1), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(4) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x2), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(6) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x3), A_C_00000000, A_C_00000000); - /* For 96kHz mode */ - /* Left ADC in. 2 of 2 */ - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0x8) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x4), A_C_00000000, A_C_00000000); - /* Right ADC in 2 of 2 */ - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xa) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x5), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xc) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x6), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xe) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x7), A_C_00000000, A_C_00000000); - /* Pavel Hofman - we still have voices, A_FXBUS2s, and - * A_P16VINs available - - * let's add 8 more capture channels - total of 16 - */ - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x10)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x8), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x12)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x9), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x14)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xa), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x16)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xb), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x18)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xc), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x1a)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xd), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x1c)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xe), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x1e)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xf), - A_C_00000000, A_C_00000000); + if (emu->card_capabilities->ca0108_chip) { + snd_printk("EMU2 inputs on\n"); + for (z = 0; z < 0x10; z++) { + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, + bit_shifter16, + A3_EMU32IN(z), + A_FXBUS2(z*2) ); + } + } else { + snd_printk("EMU inputs on\n"); + /* Capture 16 (originally 8) channels of S32_LE sound */ + + /* printk("emufx.c: gpr=0x%x, tmp=0x%x\n",gpr, tmp); */ + /* For the EMU1010: How to get 32bit values from the DSP. High 16bits into L, low 16bits into R. */ + /* A_P16VIN(0) is delayed by one sample, + * so all other A_P16VIN channels will need to also be delayed + */ + /* Left ADC in. 1 of 2 */ + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_P16VIN(0x0), A_FXBUS2(0) ); + /* Right ADC in 1 of 2 */ + gpr_map[gpr++] = 0x00000000; + /* Delaying by one sample: instead of copying the input + * value A_P16VIN to output A_FXBUS2 as in the first channel, + * we use an auxiliary register, delaying the value by one + * sample + */ + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(2) ); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x1), A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(4) ); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x2), A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(6) ); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x3), A_C_00000000, A_C_00000000); + /* For 96kHz mode */ + /* Left ADC in. 2 of 2 */ + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0x8) ); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x4), A_C_00000000, A_C_00000000); + /* Right ADC in 2 of 2 */ + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xa) ); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x5), A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xc) ); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x6), A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xe) ); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x7), A_C_00000000, A_C_00000000); + /* Pavel Hofman - we still have voices, A_FXBUS2s, and + * A_P16VINs available - + * let's add 8 more capture channels - total of 16 + */ + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x10)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x8), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x12)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x9), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x14)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xa), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x16)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xb), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x18)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xc), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x1a)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xd), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x1c)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xe), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x1e)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xf), + A_C_00000000, A_C_00000000); + } #if 0 for (z = 4; z < 8; z++) { @@ -2418,14 +2436,13 @@ static void copy_string(char *dst, char *src, char *null, int idx) strcpy(dst, src); } -static int snd_emu10k1_fx8010_info(struct snd_emu10k1 *emu, +static void snd_emu10k1_fx8010_info(struct snd_emu10k1 *emu, struct snd_emu10k1_fx8010_info *info) { char **fxbus, **extin, **extout; unsigned short fxbus_mask, extin_mask, extout_mask; int res; - memset(info, 0, sizeof(info)); info->internal_tram_size = emu->fx8010.itram_size; info->external_tram_size = emu->fx8010.etram_pages.bytes / 2; fxbus = fxbuses; @@ -2442,7 +2459,6 @@ static int snd_emu10k1_fx8010_info(struct snd_emu10k1 *emu, for (res = 16; res < 32; res++, extout++) copy_string(info->extout_names[res], extout_mask & (1 << res) ? *extout : NULL, "Unused", res); info->gpr_controls = emu->fx8010.gpr_count; - return 0; } static int snd_emu10k1_fx8010_ioctl(struct snd_hwdep * hw, struct file *file, unsigned int cmd, unsigned long arg) @@ -2463,10 +2479,7 @@ static int snd_emu10k1_fx8010_ioctl(struct snd_hwdep * hw, struct file *file, un info = kmalloc(sizeof(*info), GFP_KERNEL); if (!info) return -ENOMEM; - if ((res = snd_emu10k1_fx8010_info(emu, info)) < 0) { - kfree(info); - return res; - } + snd_emu10k1_fx8010_info(emu, info); if (copy_to_user(argp, info, sizeof(*info))) { kfree(info); return -EFAULT; diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 7b2c1dcc5337..54a2034d8edd 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz>, + * Copyright (c) by Jaroslav Kysela <perex@perex.cz>, * Takashi Iwai <tiwai@suse.de> * Creative Labs, Inc. * Routines for control of EMU10K1 chips / mixer routines @@ -400,15 +400,7 @@ static struct snd_kcontrol_new snd_emu1010_input_enum_ctls[] __devinitdata = { - -static int snd_emu1010_adc_pads_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_emu1010_adc_pads_info snd_ctl_boolean_mono_info static int snd_emu1010_adc_pads_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -456,14 +448,7 @@ static struct snd_kcontrol_new snd_emu1010_adc_pads[] __devinitdata = { EMU1010_ADC_PADS("ADC1 14dB PAD 0202 Capture Switch", EMU_HANA_0202_ADC_PAD1), }; -static int snd_emu1010_dac_pads_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_emu1010_dac_pads_info snd_ctl_boolean_mono_info static int snd_emu1010_dac_pads_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -516,17 +501,19 @@ static struct snd_kcontrol_new snd_emu1010_dac_pads[] __devinitdata = { static int snd_emu1010_internal_clock_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[2] = { - "44100", "48000" + static char *texts[4] = { + "44100", "48000", "SPDIF", "ADAT" }; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; - uinfo->value.enumerated.items = 2; - if (uinfo->value.enumerated.item > 1) - uinfo->value.enumerated.item = 1; + uinfo->value.enumerated.items = 4; + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) + uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); return 0; + + } static int snd_emu1010_internal_clock_get(struct snd_kcontrol *kcontrol, @@ -584,6 +571,44 @@ static int snd_emu1010_internal_clock_put(struct snd_kcontrol *kcontrol, /* Unmute all */ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE ); break; + + case 2: /* Take clock from S/PDIF IN */ + /* Mute all */ + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_MUTE ); + /* Default fallback clock 48kHz */ + snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, EMU_HANA_DEFCLOCK_48K ); + /* Word Clock source, sync to S/PDIF input */ + snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, + EMU_HANA_WCLOCK_HANA_SPDIF_IN | EMU_HANA_WCLOCK_1X ); + /* Set LEDs on Audio Dock */ + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, + EMU_HANA_DOCK_LEDS_2_EXT | EMU_HANA_DOCK_LEDS_2_LOCK ); + /* FIXME: We should set EMU_HANA_DOCK_LEDS_2_LOCK only when clock signal is present and valid */ + /* Allow DLL to settle */ + msleep(10); + /* Unmute all */ + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE ); + break; + + case 3: + /* Take clock from ADAT IN */ + /* Mute all */ + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_MUTE ); + /* Default fallback clock 48kHz */ + snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, EMU_HANA_DEFCLOCK_48K ); + /* Word Clock source, sync to ADAT input */ + snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, + EMU_HANA_WCLOCK_HANA_ADAT_IN | EMU_HANA_WCLOCK_1X ); + /* Set LEDs on Audio Dock */ + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, EMU_HANA_DOCK_LEDS_2_EXT | EMU_HANA_DOCK_LEDS_2_LOCK ); + /* FIXME: We should set EMU_HANA_DOCK_LEDS_2_LOCK only when clock signal is present and valid */ + /* Allow DLL to settle */ + msleep(10); + /* Unmute all */ + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE ); + + + break; } } return change; @@ -871,7 +896,7 @@ static struct snd_kcontrol_new snd_emu10k1_spdif_mask_control = .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK), - .count = 4, + .count = 3, .info = snd_emu10k1_spdif_info, .get = snd_emu10k1_spdif_get_mask }; @@ -880,7 +905,7 @@ static struct snd_kcontrol_new snd_emu10k1_spdif_control = { .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), - .count = 4, + .count = 3, .info = snd_emu10k1_spdif_info, .get = snd_emu10k1_spdif_get, .put = snd_emu10k1_spdif_put @@ -1326,14 +1351,7 @@ static struct snd_kcontrol_new snd_emu10k1_efx_attn_control = .put = snd_emu10k1_efx_attn_put }; -static int snd_emu10k1_shared_spdif_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_emu10k1_shared_spdif_info snd_ctl_boolean_mono_info static int snd_emu10k1_shared_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff --git a/sound/pci/emu10k1/emumpu401.c b/sound/pci/emu10k1/emumpu401.c index 950c6bcd6b7d..04c7cf703531 100644 --- a/sound/pci/emu10k1/emumpu401.c +++ b/sound/pci/emu10k1/emumpu401.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Routines for control of EMU10K1 MPU-401 in UART mode * * diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index eda5cb373ded..5ce5befc701b 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Creative Labs, Inc. * Routines for control of EMU10K1 chips / PCM routines * Multichannel PCM support Copyright (c) Lee Revell <rlrevell@joe-job.com> diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c index 2c1585991bc8..c3fb10e81c9e 100644 --- a/sound/pci/emu10k1/emuproc.c +++ b/sound/pci/emu10k1/emuproc.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Creative Labs, Inc. * Routines for control of EMU10K1 chips / proc interface routines * @@ -240,8 +240,42 @@ static void snd_emu10k1_proc_spdif_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_emu10k1 *emu = entry->private_data; - snd_emu10k1_proc_spdif_status(emu, buffer, "CD-ROM S/PDIF In", CDCS, CDSRCS); - snd_emu10k1_proc_spdif_status(emu, buffer, "Optical or Coax S/PDIF In", GPSCS, GPSRCS); + u32 value; + u32 value2; + unsigned long flags; + u32 rate; + + if (emu->card_capabilities->emu1010) { + spin_lock_irqsave(&emu->emu_lock, flags); + snd_emu1010_fpga_read(emu, 0x38, &value); + spin_unlock_irqrestore(&emu->emu_lock, flags); + if ((value & 0x1) == 0) { + spin_lock_irqsave(&emu->emu_lock, flags); + snd_emu1010_fpga_read(emu, 0x2a, &value); + snd_emu1010_fpga_read(emu, 0x2b, &value2); + spin_unlock_irqrestore(&emu->emu_lock, flags); + rate = 0x1770000 / (((value << 5) | value2)+1); + snd_iprintf(buffer, "ADAT Locked : %u\n", rate); + } else { + snd_iprintf(buffer, "ADAT Unlocked\n"); + } + spin_lock_irqsave(&emu->emu_lock, flags); + snd_emu1010_fpga_read(emu, 0x20, &value); + spin_unlock_irqrestore(&emu->emu_lock, flags); + if ((value & 0x4) == 0) { + spin_lock_irqsave(&emu->emu_lock, flags); + snd_emu1010_fpga_read(emu, 0x28, &value); + snd_emu1010_fpga_read(emu, 0x29, &value2); + spin_unlock_irqrestore(&emu->emu_lock, flags); + rate = 0x1770000 / (((value << 5) | value2)+1); + snd_iprintf(buffer, "SPDIF Locked : %d\n", rate); + } else { + snd_iprintf(buffer, "SPDIF Unlocked\n"); + } + } else { + snd_emu10k1_proc_spdif_status(emu, buffer, "CD-ROM S/PDIF In", CDCS, CDSRCS); + snd_emu10k1_proc_spdif_status(emu, buffer, "Optical or Coax S/PDIF In", GPSCS, GPSRCS); + } #if 0 val = snd_emu10k1_ptr_read(emu, ZVSRCS, 0); snd_iprintf(buffer, "\nZoomed Video\n"); @@ -379,20 +413,16 @@ static void snd_emu_proc_emu1010_reg_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_emu10k1 *emu = entry->private_data; - unsigned long value; + int value; unsigned long flags; - unsigned long regs; int i; snd_iprintf(buffer, "EMU1010 Registers:\n\n"); - for(i = 0; i < 0x30; i+=1) { + for(i = 0; i < 0x40; i+=1) { spin_lock_irqsave(&emu->emu_lock, flags); - regs=i+0x40; /* 0x40 upwards are registers. */ - outl(regs, emu->port + A_IOCFG); - outl(regs | 0x80, emu->port + A_IOCFG); /* High bit clocks the value into the fpga. */ - value = inl(emu->port + A_IOCFG); + snd_emu1010_fpga_read(emu, i, &value); spin_unlock_irqrestore(&emu->emu_lock, flags); - snd_iprintf(buffer, "%02X: %08lX, %02lX\n", i, value, (value >> 8) & 0x7f); + snd_iprintf(buffer, "%02X: %08X, %02X\n", i, value, (value >> 8) & 0x7f); } } @@ -555,9 +585,9 @@ int __devinit snd_emu10k1_proc_init(struct snd_emu10k1 * emu) { struct snd_info_entry *entry; #ifdef CONFIG_SND_DEBUG - if ((emu->card_capabilities->emu1010) && - snd_card_proc_new(emu->card, "emu1010_regs", &entry)) { - snd_info_set_text_ops(entry, emu, snd_emu_proc_emu1010_reg_read); + if (emu->card_capabilities->emu1010) { + if (! snd_card_proc_new(emu->card, "emu1010_regs", &entry)) + snd_info_set_text_ops(entry, emu, snd_emu_proc_emu1010_reg_read); } if (! snd_card_proc_new(emu->card, "io_regs", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_io_reg_read); diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c index 116e1c8d9361..6702c15fefa3 100644 --- a/sound/pci/emu10k1/io.c +++ b/sound/pci/emu10k1/io.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Creative Labs, Inc. * Routines for control of EMU10K1 chips * @@ -226,9 +226,9 @@ int snd_emu10k1_i2c_write(struct snd_emu10k1 *emu, return 0; } -int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, int reg, int value) +int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, u32 reg, u32 value) { - if (reg < 0 || reg > 0x3f) + if (reg > 0x3f) return 1; reg += 0x40; /* 0x40 upwards are registers. */ if (value < 0 || value > 0x3f) /* 0 to 0x3f are values */ @@ -244,9 +244,9 @@ int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, int reg, int value) return 0; } -int snd_emu1010_fpga_read(struct snd_emu10k1 * emu, int reg, int *value) +int snd_emu1010_fpga_read(struct snd_emu10k1 * emu, u32 reg, u32 *value) { - if (reg < 0 || reg > 0x3f) + if (reg > 0x3f) return 1; reg += 0x40; /* 0x40 upwards are registers. */ outl(reg, emu->port + A_IOCFG); @@ -261,7 +261,7 @@ int snd_emu1010_fpga_read(struct snd_emu10k1 * emu, int reg, int *value) /* Each Destination has one and only one Source, * but one Source can feed any number of Destinations simultaneously. */ -int snd_emu1010_fpga_link_dst_src_write(struct snd_emu10k1 * emu, int dst, int src) +int snd_emu1010_fpga_link_dst_src_write(struct snd_emu10k1 * emu, u32 dst, u32 src) { snd_emu1010_fpga_write(emu, 0x00, ((dst >> 8) & 0x3f) ); snd_emu1010_fpga_write(emu, 0x01, (dst & 0x3f) ); diff --git a/sound/pci/emu10k1/irq.c b/sound/pci/emu10k1/irq.c index 4f18f7e8bcfb..3c114b45e0b2 100644 --- a/sound/pci/emu10k1/irq.c +++ b/sound/pci/emu10k1/irq.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Creative Labs, Inc. * Routines for IRQ control of EMU10K1 chips * diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 4fcaefe5a3c5..48097c6bb15c 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Copyright (c) by Takashi Iwai <tiwai@suse.de> * * EMU10K1 memory page allocation (PTB area) diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c index 7ee19c63c2c8..d619a3842cdd 100644 --- a/sound/pci/emu10k1/p16v.c +++ b/sound/pci/emu10k1/p16v.c @@ -124,11 +124,12 @@ /* hardware definition */ static struct snd_pcm_hardware snd_p16v_playback_hw = { - .info = (SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_RESUME | - SNDRV_PCM_INFO_MMAP_VALID), + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_SYNC_START, .formats = SNDRV_PCM_FMTBIT_S32_LE, /* Only supports 24-bit samples padded to 32 bits. */ .rates = SNDRV_PCM_RATE_192000 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100, .rate_min = 44100, @@ -207,6 +208,11 @@ static int snd_p16v_pcm_open_playback_channel(struct snd_pcm_substream *substrea if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) return err; + runtime->sync.id32[0] = substream->pcm->card->number; + runtime->sync.id32[1] = 'P'; + runtime->sync.id32[2] = 16; + runtime->sync.id32[3] = 'V'; + return 0; } /* open_capture callback */ @@ -448,6 +454,9 @@ static int snd_p16v_pcm_trigger_playback(struct snd_pcm_substream *substream, break; } snd_pcm_group_for_each_entry(s, substream) { + if (snd_pcm_substream_chip(s) != emu || + s->stream != SNDRV_PCM_STREAM_PLAYBACK) + continue; runtime = s->runtime; epcm = runtime->private_data; channel = substream->pcm->device-emu->p16v_device_offset; diff --git a/sound/pci/emu10k1/voice.c b/sound/pci/emu10k1/voice.c index 1db50fe61475..04fa8492abb0 100644 --- a/sound/pci/emu10k1/voice.c +++ b/sound/pci/emu10k1/voice.c @@ -1,5 +1,5 @@ /* - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Creative Labs, Inc. * Lee Revell <rlrevell@joe-job.com> * Routines for control of EMU10K1 chips - voice manager diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 21cb4268a59b..b958f869cb13 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -1,6 +1,6 @@ /* * Driver for Ensoniq ES1370/ES1371 AudioPCI soundcard - * Copyright (c) by Jaroslav Kysela <perex@suse.cz>, + * Copyright (c) by Jaroslav Kysela <perex@perex.cz>, * Thomas Sailer <sailer@ife.ee.ethz.ch> * * This program is free software; you can redistribute it and/or modify @@ -61,7 +61,7 @@ #endif -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>, Thomas Sailer <sailer@ife.ee.ethz.ch>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, Thomas Sailer <sailer@ife.ee.ethz.ch>"); MODULE_LICENSE("GPL"); #ifdef CHIP1370 MODULE_DESCRIPTION("Ensoniq AudioPCI ES1370"); @@ -1419,15 +1419,7 @@ static int snd_ens1373_spdif_stream_put(struct snd_kcontrol *kcontrol, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .info = snd_es1371_spdif_info, \ .get = snd_es1371_spdif_get, .put = snd_es1371_spdif_put } -static int snd_es1371_spdif_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_es1371_spdif_info snd_ctl_boolean_mono_info static int snd_es1371_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1489,15 +1481,7 @@ static struct snd_kcontrol_new snd_es1371_mixer_spdif[] __devinitdata = { }; -static int snd_es1373_rear_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_es1373_rear_info snd_ctl_boolean_mono_info static int snd_es1373_rear_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1542,15 +1526,7 @@ static struct snd_kcontrol_new snd_ens1373_rear __devinitdata = .put = snd_es1373_rear_put, }; -static int snd_es1373_line_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_es1373_line_info snd_ctl_boolean_mono_info static int snd_es1373_line_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1707,15 +1683,7 @@ static int __devinit snd_ensoniq_1371_mixer(struct ensoniq *ensoniq, .get = snd_ensoniq_control_get, .put = snd_ensoniq_control_put, \ .private_value = mask } -static int snd_ensoniq_control_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_ensoniq_control_info snd_ctl_boolean_mono_info static int snd_ensoniq_control_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index fec29a108945..fb25abe68a02 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -1,7 +1,7 @@ /* * Driver for ESS Solo-1 (ES1938, ES1946, ES1969) soundcard * Copyright (c) by Jaromir Koutek <miri@punknet.cz>, - * Jaroslav Kysela <perex@suse.cz>, + * Jaroslav Kysela <perex@perex.cz>, * Thomas Sailer <sailer@ife.ee.ethz.ch>, * Abramo Bagnara <abramo@alsa-project.org>, * Markus Gruber <gruber@eikon.tum.de> @@ -1066,15 +1066,7 @@ static int snd_es1938_put_mux(struct snd_kcontrol *kcontrol, return snd_es1938_mixer_bits(chip, 0x1c, 0x07, val) != val; } -static int snd_es1938_info_spatializer_enable(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_es1938_info_spatializer_enable snd_ctl_boolean_mono_info static int snd_es1938_get_spatializer_enable(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1120,15 +1112,7 @@ static int snd_es1938_get_hw_volume(struct snd_kcontrol *kcontrol, return 0; } -static int snd_es1938_info_hw_switch(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_es1938_info_hw_switch snd_ctl_boolean_stereo_info static int snd_es1938_get_hw_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 2faf009076bb..d69b11d1f993 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -843,10 +843,9 @@ static void snd_es1968_bob_dec(struct es1968 *chip) snd_es1968_bob_stop(chip); else if (chip->bob_freq > ESM_BOB_FREQ) { /* check reduction of timer frequency */ - struct list_head *p; int max_freq = ESM_BOB_FREQ; - list_for_each(p, &chip->substream_list) { - struct esschan *es = list_entry(p, struct esschan, list); + struct esschan *es; + list_for_each_entry(es, &chip->substream_list, list) { if (max_freq < es->bob_freq) max_freq = es->bob_freq; } @@ -1316,12 +1315,11 @@ static struct snd_pcm_hardware snd_es1968_capture = { static int calc_available_memory_size(struct es1968 *chip) { - struct list_head *p; int max_size = 0; - + struct esm_memory *buf; + mutex_lock(&chip->memory_mutex); - list_for_each(p, &chip->buf_list) { - struct esm_memory *buf = list_entry(p, struct esm_memory, list); + list_for_each_entry(buf, &chip->buf_list, list) { if (buf->empty && buf->buf.bytes > max_size) max_size = buf->buf.bytes; } @@ -1335,12 +1333,10 @@ static int calc_available_memory_size(struct es1968 *chip) static struct esm_memory *snd_es1968_new_memory(struct es1968 *chip, int size) { struct esm_memory *buf; - struct list_head *p; - + size = ALIGN(size, ESM_MEM_ALIGN); mutex_lock(&chip->memory_mutex); - list_for_each(p, &chip->buf_list) { - buf = list_entry(p, struct esm_memory, list); + list_for_each_entry(buf, &chip->buf_list, list) { if (buf->empty && buf->buf.bytes >= size) goto __found; } @@ -1938,10 +1934,9 @@ static irqreturn_t snd_es1968_interrupt(int irq, void *dev_id) } if (event & ESM_SOUND_IRQ) { - struct list_head *p; + struct esschan *es; spin_lock(&chip->substream_lock); - list_for_each(p, &chip->substream_list) { - struct esschan *es = list_entry(p, struct esschan, list); + list_for_each_entry(es, &chip->substream_list, list) { if (es->running) snd_es1968_update_pcm(chip, es); } @@ -2345,7 +2340,7 @@ static int es1968_resume(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); struct es1968 *chip = card->private_data; - struct list_head *p; + struct esschan *es; if (! chip->do_pm) return 0; @@ -2374,8 +2369,7 @@ static int es1968_resume(struct pci_dev *pci) /* restore ac97 state */ snd_ac97_resume(chip->ac97); - list_for_each(p, &chip->substream_list) { - struct esschan *es = list_entry(p, struct esschan, list); + list_for_each_entry(es, &chip->substream_list, list) { switch (es->mode) { case ESM_MODE_PLAY: snd_es1968_playback_setup(chip, es, es->substream->runtime); diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 11015178e207..9939109f05a2 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1,6 +1,6 @@ /* * The driver for the ForteMedia FM801 based soundcards - * Copyright (c) by Jaroslav Kysela <perex@suse.cz> + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * * Support FM only card by Andy Shevchenko <andy@smile.org.ua> * @@ -42,7 +42,7 @@ #define TEA575X_RADIO 1 #endif -MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("ForteMedia FM801"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{ForteMedia,FM801}," diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index b2484bbdcc1d..ab0c726d648e 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -1,19 +1,18 @@ -snd-hda-intel-objs := hda_intel.o +snd-hda-intel-y := hda_intel.o # since snd-hda-intel is the only driver using hda-codec, # merge it into a single module although it was originally # designed to be individual modules -snd-hda-intel-objs += hda_codec.o \ - hda_generic.o \ - patch_realtek.o \ - patch_cmedia.o \ - patch_analog.o \ - patch_sigmatel.o \ - patch_si3054.o \ - patch_atihdmi.o \ - patch_conexant.o \ - patch_via.o -ifdef CONFIG_PROC_FS -snd-hda-intel-objs += hda_proc.o -endif +snd-hda-intel-y += hda_codec.o +snd-hda-intel-$(CONFIG_PROC_FS) += hda_proc.o +snd-hda-intel-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o +snd-hda-intel-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o +snd-hda-intel-$(CONFIG_SND_HDA_CODEC_REALTEK) += patch_realtek.o +snd-hda-intel-$(CONFIG_SND_HDA_CODEC_CMEDIA) += patch_cmedia.o +snd-hda-intel-$(CONFIG_SND_HDA_CODEC_ANALOG) += patch_analog.o +snd-hda-intel-$(CONFIG_SND_HDA_CODEC_SIGMATEL) += patch_sigmatel.o +snd-hda-intel-$(CONFIG_SND_HDA_CODEC_SI3054) += patch_si3054.o +snd-hda-intel-$(CONFIG_SND_HDA_CODEC_ATIHDMI) += patch_atihdmi.o +snd-hda-intel-$(CONFIG_SND_HDA_CODEC_CONEXANT) += patch_conexant.o +snd-hda-intel-$(CONFIG_SND_HDA_CODEC_VIA) += patch_via.o obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-intel.o diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index f87f8f088956..187533e477c6 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -31,7 +31,15 @@ #include <sound/tlv.h> #include <sound/initval.h> #include "hda_local.h" - +#include <sound/hda_hwdep.h> + +#ifdef CONFIG_SND_HDA_POWER_SAVE +/* define this option here to hide as static */ +static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; +module_param(power_save, int, 0644); +MODULE_PARM_DESC(power_save, "Automatic power-saving timeout " + "(in second, 0 = disable)."); +#endif /* * vendor / preset table @@ -59,6 +67,13 @@ static struct hda_vendor_id hda_vendor_ids[] = { #include "hda_patch.h" +#ifdef CONFIG_SND_HDA_POWER_SAVE +static void hda_power_work(struct work_struct *work); +static void hda_keep_power_on(struct hda_codec *codec); +#else +static inline void hda_keep_power_on(struct hda_codec *codec) {} +#endif + /** * snd_hda_codec_read - send a command and get the response * @codec: the HDA codec @@ -76,12 +91,14 @@ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, unsigned int verb, unsigned int parm) { unsigned int res; + snd_hda_power_up(codec); mutex_lock(&codec->bus->cmd_mutex); if (!codec->bus->ops.command(codec, nid, direct, verb, parm)) res = codec->bus->ops.get_response(codec); else res = (unsigned int)-1; mutex_unlock(&codec->bus->cmd_mutex); + snd_hda_power_down(codec); return res; } @@ -101,9 +118,11 @@ int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct, unsigned int verb, unsigned int parm) { int err; + snd_hda_power_up(codec); mutex_lock(&codec->bus->cmd_mutex); err = codec->bus->ops.command(codec, nid, direct, verb, parm); mutex_unlock(&codec->bus->cmd_mutex); + snd_hda_power_down(codec); return err; } @@ -136,6 +155,8 @@ int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid, unsigned int parm; parm = snd_hda_param_read(codec, nid, AC_PAR_NODE_COUNT); + if (parm == -1) + return 0; *start_id = (parm >> 16) & 0x7fff; return (int)(parm & 0x7fff); } @@ -387,6 +408,13 @@ int __devinit snd_hda_bus_new(struct snd_card *card, return 0; } +#ifdef CONFIG_SND_HDA_GENERIC +#define is_generic_config(codec) \ + (codec->bus->modelname && !strcmp(codec->bus->modelname, "generic")) +#else +#define is_generic_config(codec) 0 +#endif + /* * find a matching codec preset */ @@ -395,7 +423,7 @@ find_codec_preset(struct hda_codec *codec) { const struct hda_codec_preset **tbl, *preset; - if (codec->bus->modelname && !strcmp(codec->bus->modelname, "generic")) + if (is_generic_config(codec)) return NULL; /* use the generic parser */ for (tbl = hda_preset_tables; *tbl; tbl++) { @@ -486,6 +514,10 @@ static int read_widget_caps(struct hda_codec *codec, hda_nid_t fg_node) } +static void init_hda_cache(struct hda_cache_rec *cache, + unsigned int record_size); +static void free_hda_cache(struct hda_cache_rec *cache); + /* * codec destructor */ @@ -493,17 +525,20 @@ static void snd_hda_codec_free(struct hda_codec *codec) { if (!codec) return; +#ifdef CONFIG_SND_HDA_POWER_SAVE + cancel_delayed_work(&codec->power_work); + flush_scheduled_work(); +#endif list_del(&codec->list); codec->bus->caddr_tbl[codec->addr] = NULL; if (codec->patch_ops.free) codec->patch_ops.free(codec); - kfree(codec->amp_info); + free_hda_cache(&codec->amp_cache); + free_hda_cache(&codec->cmd_cache); kfree(codec->wcaps); kfree(codec); } -static void init_amp_hash(struct hda_codec *codec); - /** * snd_hda_codec_new - create a HDA codec * @bus: the bus to assign @@ -537,7 +572,17 @@ int __devinit snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, codec->bus = bus; codec->addr = codec_addr; mutex_init(&codec->spdif_mutex); - init_amp_hash(codec); + init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); + init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); + +#ifdef CONFIG_SND_HDA_POWER_SAVE + INIT_DELAYED_WORK(&codec->power_work, hda_power_work); + /* snd_hda_codec_new() marks the codec as power-up, and leave it as is. + * the caller has to power down appropriatley after initialization + * phase. + */ + hda_keep_power_on(codec); +#endif list_add_tail(&codec->list, &bus->codec_list); bus->caddr_tbl[codec_addr] = codec; @@ -581,10 +626,26 @@ int __devinit snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, snd_hda_get_codec_name(codec, bus->card->mixername, sizeof(bus->card->mixername)); - if (codec->preset && codec->preset->patch) - err = codec->preset->patch(codec); - else +#ifdef CONFIG_SND_HDA_GENERIC + if (is_generic_config(codec)) { err = snd_hda_parse_generic_codec(codec); + goto patched; + } +#endif + if (codec->preset && codec->preset->patch) { + err = codec->preset->patch(codec); + goto patched; + } + + /* call the default parser */ +#ifdef CONFIG_SND_HDA_GENERIC + err = snd_hda_parse_generic_codec(codec); +#else + printk(KERN_ERR "hda-codec: No codec parser is available\n"); + err = -ENODEV; +#endif + + patched: if (err < 0) { snd_hda_codec_free(codec); return err; @@ -594,6 +655,9 @@ int __devinit snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, init_unsol_queue(bus); snd_hda_codec_proc_new(codec); +#ifdef CONFIG_SND_HDA_HWDEP + snd_hda_create_hwdep(codec); +#endif sprintf(component, "HDA:%08x", codec->vendor_id); snd_component_add(codec->bus->card, component); @@ -637,59 +701,72 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, #define INFO_AMP_VOL(ch) (1 << (1 + (ch))) /* initialize the hash table */ -static void __devinit init_amp_hash(struct hda_codec *codec) +static void __devinit init_hda_cache(struct hda_cache_rec *cache, + unsigned int record_size) +{ + memset(cache, 0, sizeof(*cache)); + memset(cache->hash, 0xff, sizeof(cache->hash)); + cache->record_size = record_size; +} + +static void free_hda_cache(struct hda_cache_rec *cache) { - memset(codec->amp_hash, 0xff, sizeof(codec->amp_hash)); - codec->num_amp_entries = 0; - codec->amp_info_size = 0; - codec->amp_info = NULL; + kfree(cache->buffer); } /* query the hash. allocate an entry if not found. */ -static struct hda_amp_info *get_alloc_amp_hash(struct hda_codec *codec, u32 key) +static struct hda_cache_head *get_alloc_hash(struct hda_cache_rec *cache, + u32 key) { - u16 idx = key % (u16)ARRAY_SIZE(codec->amp_hash); - u16 cur = codec->amp_hash[idx]; - struct hda_amp_info *info; + u16 idx = key % (u16)ARRAY_SIZE(cache->hash); + u16 cur = cache->hash[idx]; + struct hda_cache_head *info; while (cur != 0xffff) { - info = &codec->amp_info[cur]; + info = (struct hda_cache_head *)(cache->buffer + + cur * cache->record_size); if (info->key == key) return info; cur = info->next; } /* add a new hash entry */ - if (codec->num_amp_entries >= codec->amp_info_size) { + if (cache->num_entries >= cache->size) { /* reallocate the array */ - int new_size = codec->amp_info_size + 64; - struct hda_amp_info *new_info; - new_info = kcalloc(new_size, sizeof(struct hda_amp_info), - GFP_KERNEL); - if (!new_info) { + unsigned int new_size = cache->size + 64; + void *new_buffer; + new_buffer = kcalloc(new_size, cache->record_size, GFP_KERNEL); + if (!new_buffer) { snd_printk(KERN_ERR "hda_codec: " "can't malloc amp_info\n"); return NULL; } - if (codec->amp_info) { - memcpy(new_info, codec->amp_info, - codec->amp_info_size * - sizeof(struct hda_amp_info)); - kfree(codec->amp_info); + if (cache->buffer) { + memcpy(new_buffer, cache->buffer, + cache->size * cache->record_size); + kfree(cache->buffer); } - codec->amp_info_size = new_size; - codec->amp_info = new_info; + cache->size = new_size; + cache->buffer = new_buffer; } - cur = codec->num_amp_entries++; - info = &codec->amp_info[cur]; + cur = cache->num_entries++; + info = (struct hda_cache_head *)(cache->buffer + + cur * cache->record_size); info->key = key; - info->status = 0; /* not initialized yet */ - info->next = codec->amp_hash[idx]; - codec->amp_hash[idx] = cur; + info->val = 0; + info->next = cache->hash[idx]; + cache->hash[idx] = cur; return info; } +/* query and allocate an amp hash entry */ +static inline struct hda_amp_info * +get_alloc_amp_hash(struct hda_codec *codec, u32 key) +{ + return (struct hda_amp_info *)get_alloc_hash(&codec->amp_cache, key); +} + /* * query AMP capabilities for the given widget and direction */ @@ -700,7 +777,7 @@ static u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction) info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, 0)); if (!info) return 0; - if (!(info->status & INFO_AMP_CAPS)) { + if (!(info->head.val & INFO_AMP_CAPS)) { if (!(get_wcaps(codec, nid) & AC_WCAP_AMP_OVRD)) nid = codec->afg; info->amp_caps = snd_hda_param_read(codec, nid, @@ -708,7 +785,7 @@ static u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction) AC_PAR_AMP_OUT_CAP : AC_PAR_AMP_IN_CAP); if (info->amp_caps) - info->status |= INFO_AMP_CAPS; + info->head.val |= INFO_AMP_CAPS; } return info->amp_caps; } @@ -722,7 +799,7 @@ int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, if (!info) return -EINVAL; info->amp_caps = caps; - info->status |= INFO_AMP_CAPS; + info->head.val |= INFO_AMP_CAPS; return 0; } @@ -736,7 +813,7 @@ static unsigned int get_vol_mute(struct hda_codec *codec, { u32 val, parm; - if (info->status & INFO_AMP_VOL(ch)) + if (info->head.val & INFO_AMP_VOL(ch)) return info->vol[ch]; parm = ch ? AC_AMP_GET_RIGHT : AC_AMP_GET_LEFT; @@ -745,7 +822,7 @@ static unsigned int get_vol_mute(struct hda_codec *codec, val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_AMP_GAIN_MUTE, parm); info->vol[ch] = val & 0xff; - info->status |= INFO_AMP_VOL(ch); + info->head.val |= INFO_AMP_VOL(ch); return info->vol[ch]; } @@ -792,12 +869,50 @@ int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, return 0; val &= mask; val |= get_vol_mute(codec, info, nid, ch, direction, idx) & ~mask; - if (info->vol[ch] == val && !codec->in_resume) + if (info->vol[ch] == val) return 0; put_vol_mute(codec, info, nid, ch, direction, idx, val); return 1; } +/* + * update the AMP stereo with the same mask and value + */ +int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid, + int direction, int idx, int mask, int val) +{ + int ch, ret = 0; + for (ch = 0; ch < 2; ch++) + ret |= snd_hda_codec_amp_update(codec, nid, ch, direction, + idx, mask, val); + return ret; +} + +#ifdef SND_HDA_NEEDS_RESUME +/* resume the all amp commands from the cache */ +void snd_hda_codec_resume_amp(struct hda_codec *codec) +{ + struct hda_amp_info *buffer = codec->amp_cache.buffer; + int i; + + for (i = 0; i < codec->amp_cache.size; i++, buffer++) { + u32 key = buffer->head.key; + hda_nid_t nid; + unsigned int idx, dir, ch; + if (!key) + continue; + nid = key & 0xff; + idx = (key >> 16) & 0xff; + dir = (key >> 24) & 0xff; + for (ch = 0; ch < 2; ch++) { + if (!(buffer->head.val & INFO_AMP_VOL(ch))) + continue; + put_vol_mute(codec, buffer, nid, ch, dir, idx, + buffer->vol[ch]); + } + } +} +#endif /* SND_HDA_NEEDS_RESUME */ /* * AMP control callbacks @@ -844,9 +959,11 @@ int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol, long *valp = ucontrol->value.integer.value; if (chs & 1) - *valp++ = snd_hda_codec_amp_read(codec, nid, 0, dir, idx) & 0x7f; + *valp++ = snd_hda_codec_amp_read(codec, nid, 0, dir, idx) + & HDA_AMP_VOLMASK; if (chs & 2) - *valp = snd_hda_codec_amp_read(codec, nid, 1, dir, idx) & 0x7f; + *valp = snd_hda_codec_amp_read(codec, nid, 1, dir, idx) + & HDA_AMP_VOLMASK; return 0; } @@ -861,6 +978,7 @@ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol, long *valp = ucontrol->value.integer.value; int change = 0; + snd_hda_power_up(codec); if (chs & 1) { change = snd_hda_codec_amp_update(codec, nid, 0, dir, idx, 0x7f, *valp); @@ -869,6 +987,7 @@ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol, if (chs & 2) change |= snd_hda_codec_amp_update(codec, nid, 1, dir, idx, 0x7f, *valp); + snd_hda_power_down(codec); return change; } @@ -923,10 +1042,10 @@ int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol, if (chs & 1) *valp++ = (snd_hda_codec_amp_read(codec, nid, 0, dir, idx) & - 0x80) ? 0 : 1; + HDA_AMP_MUTE) ? 0 : 1; if (chs & 2) *valp = (snd_hda_codec_amp_read(codec, nid, 1, dir, idx) & - 0x80) ? 0 : 1; + HDA_AMP_MUTE) ? 0 : 1; return 0; } @@ -941,15 +1060,22 @@ int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, long *valp = ucontrol->value.integer.value; int change = 0; + snd_hda_power_up(codec); if (chs & 1) { change = snd_hda_codec_amp_update(codec, nid, 0, dir, idx, - 0x80, *valp ? 0 : 0x80); + HDA_AMP_MUTE, + *valp ? 0 : HDA_AMP_MUTE); valp++; } if (chs & 2) change |= snd_hda_codec_amp_update(codec, nid, 1, dir, idx, - 0x80, *valp ? 0 : 0x80); - + HDA_AMP_MUTE, + *valp ? 0 : HDA_AMP_MUTE); +#ifdef CONFIG_SND_HDA_POWER_SAVE + if (codec->patch_ops.check_power_status) + codec->patch_ops.check_power_status(codec, nid); +#endif + snd_hda_power_down(codec); return change; } @@ -1002,6 +1128,93 @@ int snd_hda_mixer_bind_switch_put(struct snd_kcontrol *kcontrol, } /* + * generic bound volume/swtich controls + */ +int snd_hda_mixer_bind_ctls_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hda_bind_ctls *c; + int err; + + c = (struct hda_bind_ctls *)kcontrol->private_value; + mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + kcontrol->private_value = *c->values; + err = c->ops->info(kcontrol, uinfo); + kcontrol->private_value = (long)c; + mutex_unlock(&codec->spdif_mutex); + return err; +} + +int snd_hda_mixer_bind_ctls_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hda_bind_ctls *c; + int err; + + c = (struct hda_bind_ctls *)kcontrol->private_value; + mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + kcontrol->private_value = *c->values; + err = c->ops->get(kcontrol, ucontrol); + kcontrol->private_value = (long)c; + mutex_unlock(&codec->spdif_mutex); + return err; +} + +int snd_hda_mixer_bind_ctls_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hda_bind_ctls *c; + unsigned long *vals; + int err = 0, change = 0; + + c = (struct hda_bind_ctls *)kcontrol->private_value; + mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + for (vals = c->values; *vals; vals++) { + kcontrol->private_value = *vals; + err = c->ops->put(kcontrol, ucontrol); + if (err < 0) + break; + change |= err; + } + kcontrol->private_value = (long)c; + mutex_unlock(&codec->spdif_mutex); + return err < 0 ? err : change; +} + +int snd_hda_mixer_bind_tlv(struct snd_kcontrol *kcontrol, int op_flag, + unsigned int size, unsigned int __user *tlv) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hda_bind_ctls *c; + int err; + + c = (struct hda_bind_ctls *)kcontrol->private_value; + mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + kcontrol->private_value = *c->values; + err = c->ops->tlv(kcontrol, op_flag, size, tlv); + kcontrol->private_value = (long)c; + mutex_unlock(&codec->spdif_mutex); + return err; +} + +struct hda_ctl_ops snd_hda_bind_vol = { + .info = snd_hda_mixer_amp_volume_info, + .get = snd_hda_mixer_amp_volume_get, + .put = snd_hda_mixer_amp_volume_put, + .tlv = snd_hda_mixer_amp_tlv +}; + +struct hda_ctl_ops snd_hda_bind_sw = { + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = snd_hda_mixer_amp_switch_put, + .tlv = snd_hda_mixer_amp_tlv +}; + +/* * SPDIF out controls */ @@ -1118,26 +1331,20 @@ static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol, change = codec->spdif_ctls != val; codec->spdif_ctls = val; - if (change || codec->in_resume) { - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, - val & 0xff); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_2, - val >> 8); + if (change) { + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_DIGI_CONVERT_1, + val & 0xff); + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_DIGI_CONVERT_2, + val >> 8); } mutex_unlock(&codec->spdif_mutex); return change; } -static int snd_hda_spdif_out_switch_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_hda_spdif_out_switch_info snd_ctl_boolean_mono_info static int snd_hda_spdif_out_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1161,17 +1368,16 @@ static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kcontrol, if (ucontrol->value.integer.value[0]) val |= AC_DIG1_ENABLE; change = codec->spdif_ctls != val; - if (change || codec->in_resume) { + if (change) { codec->spdif_ctls = val; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, - val & 0xff); + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_DIGI_CONVERT_1, + val & 0xff); /* unmute amp switch (if any) */ if ((get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) && (val & AC_DIG1_ENABLE)) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | - AC_AMP_SET_OUTPUT); + snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, + HDA_AMP_MUTE, 0); } mutex_unlock(&codec->spdif_mutex); return change; @@ -1219,8 +1425,7 @@ static struct snd_kcontrol_new dig_mixes[] = { * * Returns 0 if successful, or a negative error code. */ -int __devinit snd_hda_create_spdif_out_ctls(struct hda_codec *codec, - hda_nid_t nid) +int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid) { int err; struct snd_kcontrol *kctl; @@ -1264,10 +1469,10 @@ static int snd_hda_spdif_in_switch_put(struct snd_kcontrol *kcontrol, mutex_lock(&codec->spdif_mutex); change = codec->spdif_in_enable != val; - if (change || codec->in_resume) { + if (change) { codec->spdif_in_enable = val; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, - val); + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_DIGI_CONVERT_1, val); } mutex_unlock(&codec->spdif_mutex); return change; @@ -1318,8 +1523,7 @@ static struct snd_kcontrol_new dig_in_ctls[] = { * * Returns 0 if successful, or a negative error code. */ -int __devinit snd_hda_create_spdif_in_ctls(struct hda_codec *codec, - hda_nid_t nid) +int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid) { int err; struct snd_kcontrol *kctl; @@ -1338,6 +1542,79 @@ int __devinit snd_hda_create_spdif_in_ctls(struct hda_codec *codec, return 0; } +#ifdef SND_HDA_NEEDS_RESUME +/* + * command cache + */ + +/* build a 32bit cache key with the widget id and the command parameter */ +#define build_cmd_cache_key(nid, verb) ((verb << 8) | nid) +#define get_cmd_cache_nid(key) ((key) & 0xff) +#define get_cmd_cache_cmd(key) (((key) >> 8) & 0xffff) + +/** + * snd_hda_codec_write_cache - send a single command with caching + * @codec: the HDA codec + * @nid: NID to send the command + * @direct: direct flag + * @verb: the verb to send + * @parm: the parameter for the verb + * + * Send a single command without waiting for response. + * + * Returns 0 if successful, or a negative error code. + */ +int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, + int direct, unsigned int verb, unsigned int parm) +{ + int err; + snd_hda_power_up(codec); + mutex_lock(&codec->bus->cmd_mutex); + err = codec->bus->ops.command(codec, nid, direct, verb, parm); + if (!err) { + struct hda_cache_head *c; + u32 key = build_cmd_cache_key(nid, verb); + c = get_alloc_hash(&codec->cmd_cache, key); + if (c) + c->val = parm; + } + mutex_unlock(&codec->bus->cmd_mutex); + snd_hda_power_down(codec); + return err; +} + +/* resume the all commands from the cache */ +void snd_hda_codec_resume_cache(struct hda_codec *codec) +{ + struct hda_cache_head *buffer = codec->cmd_cache.buffer; + int i; + + for (i = 0; i < codec->cmd_cache.size; i++, buffer++) { + u32 key = buffer->key; + if (!key) + continue; + snd_hda_codec_write(codec, get_cmd_cache_nid(key), 0, + get_cmd_cache_cmd(key), buffer->val); + } +} + +/** + * snd_hda_sequence_write_cache - sequence writes with caching + * @codec: the HDA codec + * @seq: VERB array to send + * + * Send the commands sequentially from the given array. + * Thte commands are recorded on cache for power-save and resume. + * The array must be terminated with NID=0. + */ +void snd_hda_sequence_write_cache(struct hda_codec *codec, + const struct hda_verb *seq) +{ + for (; seq->nid; seq++) + snd_hda_codec_write_cache(codec, seq->nid, 0, seq->verb, + seq->param); +} +#endif /* SND_HDA_NEEDS_RESUME */ /* * set power state of the codec @@ -1345,23 +1622,86 @@ int __devinit snd_hda_create_spdif_in_ctls(struct hda_codec *codec, static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state) { - hda_nid_t nid, nid_start; - int nodes; + hda_nid_t nid; + int i; snd_hda_codec_write(codec, fg, 0, AC_VERB_SET_POWER_STATE, power_state); - nodes = snd_hda_get_sub_nodes(codec, fg, &nid_start); - for (nid = nid_start; nid < nodes + nid_start; nid++) { - if (get_wcaps(codec, nid) & AC_WCAP_POWER) + nid = codec->start_nid; + for (i = 0; i < codec->num_nodes; i++, nid++) { + if (get_wcaps(codec, nid) & AC_WCAP_POWER) { + unsigned int pincap; + /* + * don't power down the widget if it controls eapd + * and EAPD_BTLENABLE is set. + */ + pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + if (pincap & AC_PINCAP_EAPD) { + int eapd = snd_hda_codec_read(codec, nid, + 0, AC_VERB_GET_EAPD_BTLENABLE, 0); + eapd &= 0x02; + if (power_state == AC_PWRST_D3 && eapd) + continue; + } snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, power_state); + } } - if (power_state == AC_PWRST_D0) + if (power_state == AC_PWRST_D0) { + unsigned long end_time; + int state; msleep(10); + /* wait until the codec reachs to D0 */ + end_time = jiffies + msecs_to_jiffies(500); + do { + state = snd_hda_codec_read(codec, fg, 0, + AC_VERB_GET_POWER_STATE, 0); + if (state == power_state) + break; + msleep(1); + } while (time_after_eq(end_time, jiffies)); + } +} + +#ifdef SND_HDA_NEEDS_RESUME +/* + * call suspend and power-down; used both from PM and power-save + */ +static void hda_call_codec_suspend(struct hda_codec *codec) +{ + if (codec->patch_ops.suspend) + codec->patch_ops.suspend(codec, PMSG_SUSPEND); + hda_set_power_state(codec, + codec->afg ? codec->afg : codec->mfg, + AC_PWRST_D3); +#ifdef CONFIG_SND_HDA_POWER_SAVE + cancel_delayed_work(&codec->power_work); + codec->power_on = 0; + codec->power_transition = 0; +#endif +} + +/* + * kick up codec; used both from PM and power-save + */ +static void hda_call_codec_resume(struct hda_codec *codec) +{ + hda_set_power_state(codec, + codec->afg ? codec->afg : codec->mfg, + AC_PWRST_D0); + if (codec->patch_ops.resume) + codec->patch_ops.resume(codec); + else { + if (codec->patch_ops.init) + codec->patch_ops.init(codec); + snd_hda_codec_resume_amp(codec); + snd_hda_codec_resume_cache(codec); + } } +#endif /* SND_HDA_NEEDS_RESUME */ /** @@ -1376,28 +1716,24 @@ int __devinit snd_hda_build_controls(struct hda_bus *bus) { struct hda_codec *codec; - /* build controls */ list_for_each_entry(codec, &bus->codec_list, list) { - int err; - if (!codec->patch_ops.build_controls) - continue; - err = codec->patch_ops.build_controls(codec); - if (err < 0) - return err; - } - - /* initialize */ - list_for_each_entry(codec, &bus->codec_list, list) { - int err; + int err = 0; + /* fake as if already powered-on */ + hda_keep_power_on(codec); + /* then fire up */ hda_set_power_state(codec, codec->afg ? codec->afg : codec->mfg, AC_PWRST_D0); - if (!codec->patch_ops.init) - continue; - err = codec->patch_ops.init(codec); + /* continue to initialize... */ + if (codec->patch_ops.init) + err = codec->patch_ops.init(codec); + if (!err && codec->patch_ops.build_controls) + err = codec->patch_ops.build_controls(codec); + snd_hda_power_down(codec); if (err < 0) return err; } + return 0; } @@ -1789,9 +2125,9 @@ int __devinit snd_hda_build_pcms(struct hda_bus *bus) * * If no entries are matching, the function returns a negative value. */ -int __devinit snd_hda_check_board_config(struct hda_codec *codec, - int num_configs, const char **models, - const struct snd_pci_quirk *tbl) +int snd_hda_check_board_config(struct hda_codec *codec, + int num_configs, const char **models, + const struct snd_pci_quirk *tbl) { if (codec->bus->modelname && models) { int i; @@ -1841,10 +2177,9 @@ int __devinit snd_hda_check_board_config(struct hda_codec *codec, * * Returns 0 if successful, or a negative error code. */ -int __devinit snd_hda_add_new_ctls(struct hda_codec *codec, - struct snd_kcontrol_new *knew) +int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew) { - int err; + int err; for (; knew->name; knew++) { struct snd_kcontrol *kctl; @@ -1867,6 +2202,93 @@ int __devinit snd_hda_add_new_ctls(struct hda_codec *codec, return 0; } +#ifdef CONFIG_SND_HDA_POWER_SAVE +static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state); + +static void hda_power_work(struct work_struct *work) +{ + struct hda_codec *codec = + container_of(work, struct hda_codec, power_work.work); + + if (!codec->power_on || codec->power_count) { + codec->power_transition = 0; + return; + } + + hda_call_codec_suspend(codec); + if (codec->bus->ops.pm_notify) + codec->bus->ops.pm_notify(codec); +} + +static void hda_keep_power_on(struct hda_codec *codec) +{ + codec->power_count++; + codec->power_on = 1; +} + +void snd_hda_power_up(struct hda_codec *codec) +{ + codec->power_count++; + if (codec->power_on || codec->power_transition) + return; + + codec->power_on = 1; + if (codec->bus->ops.pm_notify) + codec->bus->ops.pm_notify(codec); + hda_call_codec_resume(codec); + cancel_delayed_work(&codec->power_work); + codec->power_transition = 0; +} + +void snd_hda_power_down(struct hda_codec *codec) +{ + --codec->power_count; + if (!codec->power_on || codec->power_count || codec->power_transition) + return; + if (power_save) { + codec->power_transition = 1; /* avoid reentrance */ + schedule_delayed_work(&codec->power_work, + msecs_to_jiffies(power_save * 1000)); + } +} + +int snd_hda_check_amp_list_power(struct hda_codec *codec, + struct hda_loopback_check *check, + hda_nid_t nid) +{ + struct hda_amp_list *p; + int ch, v; + + if (!check->amplist) + return 0; + for (p = check->amplist; p->nid; p++) { + if (p->nid == nid) + break; + } + if (!p->nid) + return 0; /* nothing changed */ + + for (p = check->amplist; p->nid; p++) { + for (ch = 0; ch < 2; ch++) { + v = snd_hda_codec_amp_read(codec, p->nid, ch, p->dir, + p->idx); + if (!(v & HDA_AMP_MUTE) && v > 0) { + if (!check->power_on) { + check->power_on = 1; + snd_hda_power_up(codec); + } + return 1; + } + } + } + if (check->power_on) { + check->power_on = 0; + snd_hda_power_down(codec); + } + return 0; +} +#endif /* * Channel mode helper @@ -1913,12 +2335,12 @@ int snd_hda_ch_mode_put(struct hda_codec *codec, mode = ucontrol->value.enumerated.item[0]; snd_assert(mode < num_chmodes, return -EINVAL); - if (*max_channelsp == chmode[mode].channels && !codec->in_resume) + if (*max_channelsp == chmode[mode].channels) return 0; /* change the current channel setting */ *max_channelsp = chmode[mode].channels; if (chmode[mode].sequence) - snd_hda_sequence_write(codec, chmode[mode].sequence); + snd_hda_sequence_write_cache(codec, chmode[mode].sequence); return 1; } @@ -1933,6 +2355,8 @@ int snd_hda_input_mux_info(const struct hda_input_mux *imux, uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = imux->num_items; + if (!imux->num_items) + return 0; index = uinfo->value.enumerated.item; if (index >= imux->num_items) index = imux->num_items - 1; @@ -1948,13 +2372,15 @@ int snd_hda_input_mux_put(struct hda_codec *codec, { unsigned int idx; + if (!imux->num_items) + return 0; idx = ucontrol->value.enumerated.item[0]; if (idx >= imux->num_items) idx = imux->num_items - 1; - if (*cur_val == idx && !codec->in_resume) + if (*cur_val == idx) return 0; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, - imux->items[idx].index); + snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, + imux->items[idx].index); *cur_val = idx; return 1; } @@ -2118,7 +2544,7 @@ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, * Helper for automatic ping configuration */ -static int __devinit is_in_nid_list(hda_nid_t nid, hda_nid_t *list) +static int is_in_nid_list(hda_nid_t nid, hda_nid_t *list) { for (; *list; list++) if (*list == nid) @@ -2169,9 +2595,9 @@ static void sort_pins_by_sequence(hda_nid_t * pins, short * sequences, * The digital input/output pins are assigned to dig_in_pin and dig_out_pin, * respectively. */ -int __devinit snd_hda_parse_pin_def_config(struct hda_codec *codec, - struct auto_pin_cfg *cfg, - hda_nid_t *ignore_nids) +int snd_hda_parse_pin_def_config(struct hda_codec *codec, + struct auto_pin_cfg *cfg, + hda_nid_t *ignore_nids) { hda_nid_t nid, nid_start; |