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Merge branch 'linus' of git://git.kernel.org/pub/scm/linux/kernel/git/perex/alsa 

* 'linus' of git://git.kernel.org/pub/scm/linux/kernel/git/perex/alsa: (299 commits)
  [ALSA] version 1.0.16rc2
  [ALSA] hda: fix Mic in as output
  [ALSA] emu10k1 - Another EMU0404 Board ID
  [ALSA] emu10k1 - Fix kthread handling at resume
  [ALSA] emu10k1: General cleanup, add new locks, fix alsa bug#3501, kernel bug#9304.
  [ALSA] emu10k1 - Use enum for emu_model types
  [ALSA] emu10k1 - Don't create emu1010 controls for non-emu boards
  [ALSA] emu10k1 - 1616(M) cardbus improvements
  [ALSA] snd:emu10k1: E-Mu updates. Fixes to firmware loading and support for 0404.
  [ALSA] emu10k1: Add comments regarding E-Mu ins and outs.
  [ALSA] oxygen: revert SPI clock frequency change for AK4396/WM8785
  [ALSA] es1938 - improve capture hw pointer reads
  [ALSA] HDA-Intel - Add support for Intel SCH
  [ALSA] hda: Add GPIO mute support to STAC9205
  [ALSA] hda-codec - Add Dell T3400 support
  [ALSA] hda-codec - Add model for HP DV9553EG laptop
  [ALSA] hda-codec - Control SPDIF as slave
  [ALSA] hda_intel: ALSA HD Audio patch for Intel ICH10 DeviceID's
  [ALSA] Fix Oops with PCM OSS sync
  [ALSA] hda-codec - Add speaker automute to ALC262 HP models
  ...
This commit is contained in:
Linus Torvalds 2008-02-01 10:16:28 +11:00
parent fcc3ff4f9d 2ecba4ffbb
commit e1a9c9872d
460 changed files with 24428 additions and 9877 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

210
Documentation/sound/alsa/ALSA-Configuration.txt
View File

@ -57,7 +57,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
- Default: 1
- For auto-loading more than one card, specify this
option together with snd-card-X aliases.
slots - Reserve the slot index for the given driver.
This option takes multiple strings.
See "Module Autoloading Support" section for details.
Module snd-pcm-oss
------------------
@ -148,13 +150,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
Module for sound cards based on Analog Devices AD1816A/AD1815 ISA chips.
port - port # for AD1816A chip (PnP setup)
mpu_port - port # for MPU-401 UART (PnP setup)
fm_port - port # for OPL3 (PnP setup)
irq - IRQ # for AD1816A chip (PnP setup)
mpu_irq - IRQ # for MPU-401 UART (PnP setup)
dma1 - first DMA # for AD1816A chip (PnP setup)
dma2 - second DMA # for AD1816A chip (PnP setup)
clockfreq - Clock frequency for AD1816A chip (default = 0, 33000Hz)
This module supports multiple cards, autoprobe and PnP.
@ -201,14 +196,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
Module for sound cards based on Avance Logic ALS100/ALS120 ISA chips.
port - port # for ALS100 (SB16) chip (PnP setup)
irq - IRQ # for ALS100 (SB16) chip (PnP setup)
dma8 - 8-bit DMA # for ALS100 (SB16) chip (PnP setup)
dma16 - 16-bit DMA # for ALS100 (SB16) chip (PnP setup)
mpu_port - port # for MPU-401 UART (PnP setup)
mpu_irq - IRQ # for MPU-401 (PnP setup)
fm_port - port # for OPL3 FM (PnP setup)
This module supports multiple cards, autoprobe and PnP.
The power-management is supported.
@ -302,15 +289,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
Module for sound cards based on Aztech System AZT2320 ISA chip (PnP only).
port - port # for AZT2320 chip (PnP setup)
wss_port - port # for WSS (PnP setup)
mpu_port - port # for MPU-401 UART (PnP setup)
fm_port - FM port # for AZT2320 chip (PnP setup)
irq - IRQ # for AZT2320 (WSS) chip (PnP setup)
mpu_irq - IRQ # for MPU-401 UART (PnP setup)
dma1 - 1st DMA # for AZT2320 (WSS) chip (PnP setup)
dma2 - 2nd DMA # for AZT2320 (WSS) chip (PnP setup)
This module supports multiple cards, PnP and autoprobe.
The power-management is supported.
@ -350,6 +328,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
Module for sound cards based on C-Media CMI8330 ISA chips.
isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
with isapnp=0, the following options are available:
wssport - port # for CMI8330 chip (WSS)
wssirq - IRQ # for CMI8330 chip (WSS)
wssdma - first DMA # for CMI8330 chip (WSS)
@ -404,6 +386,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
Module for sound cards based on CS4232/CS4232A ISA chips.
isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
with isapnp=0, the following options are available:
port - port # for CS4232 chip (PnP setup - 0x534)
cport - control port # for CS4232 chip (PnP setup - 0x120,0x210,0xf00)
mpu_port - port # for MPU-401 UART (PnP setup - 0x300), -1 = disable
@ -412,10 +398,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
mpu_irq - IRQ # for MPU-401 UART (9,11,12,15)
dma1 - first DMA # for CS4232 chip (0,1,3)
dma2 - second DMA # for Yamaha CS4232 chip (0,1,3), -1 = disable
isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
This module supports multiple cards. This module does not support autoprobe
thus main port must be specified!!! Other ports are optional.
(if ISA PnP is not used) thus main port must be specified!!! Other ports are
optional.
The power-management is supported.
@ -425,6 +411,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
Module for sound cards based on CS4235/CS4236/CS4236B/CS4237B/
CS4238B/CS4239 ISA chips.
isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
with isapnp=0, the following options are available:
port - port # for CS4236 chip (PnP setup - 0x534)
cport - control port # for CS4236 chip (PnP setup - 0x120,0x210,0xf00)
mpu_port - port # for MPU-401 UART (PnP setup - 0x300), -1 = disable
@ -433,7 +423,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
mpu_irq - IRQ # for MPU-401 UART (9,11,12,15)
dma1 - first DMA # for CS4236 chip (0,1,3)
dma2 - second DMA # for CS4236 chip (0,1,3), -1 = disable
isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
This module supports multiple cards. This module does not support autoprobe
(if ISA PnP is not used) thus main port and control port must be
@ -503,13 +492,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
Module for Diamond Technologies DT-019X / Avance Logic ALS-007 (PnP
only)
port - Port # (PnP setup)
mpu_port - Port # for MPU-401 (PnP setup)
fm_port - Port # for FM OPL-3 (PnP setup)
irq - IRQ # (PnP setup)
mpu_irq - IRQ # for MPU-401 (PnP setup)
dma8 - DMA # (PnP setup)
This module supports multiple cards. This module is enabled only with
ISA PnP support.
@ -607,10 +589,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
Module for sound cards based on ESS ES968 chip (PnP only).
port - port # for ES968 (SB8) chip (PnP setup)
irq - IRQ # for ES968 (SB8) chip (PnP setup)
dma1 - DMA # for ES968 (SB8) chip (PnP setup)
This module supports multiple cards, PnP and autoprobe.
The power-management is supported.
@ -633,13 +611,16 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
Module for ESS AudioDrive ES-18xx sound cards.
isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
with isapnp=0, the following options are available:
port - port # for ES-18xx chip (0x220,0x240,0x260)
mpu_port - port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable (default)
fm_port - port # for FM (optional, not used)
irq - IRQ # for ES-18xx chip (5,7,9,10)
dma1 - first DMA # for ES-18xx chip (0,1,3)
dma2 - first DMA # for ES-18xx chip (0,1,3)
isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
This module supports multiple cards, ISA PnP and autoprobe (without MPU-401
port if native ISA PnP routines are not used).
@ -763,9 +744,12 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
VIA VT8251/VT8237A,
SIS966, ULI M5461
[Multiple options for each card instance]
model - force the model name
position_fix - Fix DMA pointer (0 = auto, 1 = none, 2 = POSBUF, 3 = FIFO size)
probe_mask - Bitmask to probe codecs (default = -1, meaning all slots)
[Single (global) options]
single_cmd - Use single immediate commands to communicate with
codecs (for debugging only)
enable_msi - Enable Message Signaled Interrupt (MSI) (default = off)
@ -774,8 +758,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
power_save_controller - Reset HD-audio controller in power-saving mode
(default = on)
This module supports one card and autoprobe.
This module supports multiple cards and autoprobe.
Each codec may have a model table for different configurations.
If your machine isn't listed there, the default (usually minimal)
configuration is set up. You can pass "model=<name>" option to
@ -817,17 +801,23 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
will Will laptops (PB V7900)
replacer Replacer 672V
basic fixed pin assignment (old default model)
test for testing/debugging purpose, almost all controls can
adjusted. Appearing only when compiled with
$CONFIG_SND_DEBUG=y
auto auto-config reading BIOS (default)
ALC262
fujitsu Fujitsu Laptop
hp-bpc HP xw4400/6400/8400/9400 laptops
hp-bpc-d7000 HP BPC D7000
hp-tc-t5735 HP Thin Client T5735
hp-rp5700 HP RP5700
benq Benq ED8
benq-t31 Benq T31
hippo Hippo (ATI) with jack detection, Sony UX-90s
hippo_1 Hippo (Benq) with jack detection
sony-assamd Sony ASSAMD
ultra Samsung Q1 Ultra Vista model
basic fixed pin assignment w/o SPDIF
auto auto-config reading BIOS (default)
@ -835,6 +825,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
3stack 3-stack model
toshiba Toshiba A205
acer Acer laptops
dell Dell OEM laptops (Vostro 1200)
test for testing/debugging purpose, almost all controls can
adjusted. Appearing only when compiled with
$CONFIG_SND_DEBUG=y
auto auto-config reading BIOS (default)
ALC662
@ -843,6 +837,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
3stack-6ch-dig 3-stack (6-channel) with SPDIF
6stack-dig 6-stack with SPDIF
lenovo-101e Lenovo laptop
eeepc-p701 ASUS Eeepc P701
eeepc-ep20 ASUS Eeepc EP20
auto auto-config reading BIOS (default)
ALC882/885
@ -877,6 +873,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
haier-w66 Haier W66
6stack-hp HP machines with 6stack (Nettle boards)
3stack-hp HP machines with 3stack (Lucknow, Samba boards)
6stack-dell Dell machines with 6stack (Inspiron 530)
mitac Mitac 8252D
auto auto-config reading BIOS (default)
ALC861/660
@ -928,6 +926,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
AD1984
basic default configuration
thinkpad Lenovo Thinkpad T61/X61
dell Dell T3400
AD1986A
6stack 6-jack, separate surrounds (default)
@ -947,7 +946,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
auto auto-config reading BIOS (default)
Conexant 5045
laptop Laptop config
laptop-hpsense Laptop with HP sense (old model laptop)
laptop-micsense Laptop with Mic sense (old model fujitsu)
laptop-hpmicsense Laptop with HP and Mic senses
benq Benq R55E
test for testing/debugging purpose, almost all controls
can be adjusted. Appearing only when compiled with
$CONFIG_SND_DEBUG=y
@ -960,6 +962,10 @@ 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
Conexant 5051
laptop Basic Laptop config (default)
hp HP Spartan laptop
STAC9200
ref Reference board
dell-d21 Dell (unknown)
@ -1091,6 +1097,15 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
See hdspm.txt for details.
Module snd-hifier
-----------------
Module for the MediaTek/TempoTec HiFier Fantasia sound card.
This module supports autoprobe and multiple cards.
Power management is _not_ supported.
Module snd-ice1712
------------------
@ -1156,11 +1171,14 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
* Chaintech 9CJS
* Chaintech AV-710
* Shuttle SN25P
* Onkyo SE-90PCI
* Onkyo SE-200PCI
model - Use the given board model, one of the following:
revo51, revo71, amp2000, prodigy71, prodigy71lt,
prodigy192, aureon51, aureon71, universe, ap192,
k8x800, phase22, phase28, ms300, av710
k8x800, phase22, phase28, ms300, av710, se200pci,
se90pci
This module supports multiple cards and autoprobe.
@ -1257,15 +1275,19 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
Module for Gravis UltraSound PnP, Dynasonic 3-D/Pro, STB Sound Rage 32
and other sound cards based on AMD InterWave (tm) chip.
port - port # for InterWave chip (0x210,0x220,0x230,0x240,0x250,0x260)
irq - IRQ # for InterWave chip (3,5,9,11,12,15)
dma1 - DMA # for InterWave chip (0,1,3,5,6,7)
dma2 - DMA # for InterWave chip (0,1,3,5,6,7,-1=disable)
joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V)
midi - 1 = MIDI UART enable, 0 = MIDI UART disable (default)
pcm_voices - reserved PCM voices for the synthesizer (default 2)
effect - 1 = InterWave effects enable (default 0);
requires 8 voices
isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
with isapnp=0, the following options are available:
port - port # for InterWave chip (0x210,0x220,0x230,0x240,0x250,0x260)
irq - IRQ # for InterWave chip (3,5,9,11,12,15)
dma1 - DMA # for InterWave chip (0,1,3,5,6,7)
dma2 - DMA # for InterWave chip (0,1,3,5,6,7,-1=disable)
This module supports multiple cards, autoprobe and ISA PnP.
@ -1276,16 +1298,20 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
and other sound cards based on AMD InterWave (tm) chip with TEA6330T
circuit for extended control of bass, treble and master volume.
port - port # for InterWave chip (0x210,0x220,0x230,0x240,0x250,0x260)
port_tc - tone control (i2c bus) port # for TEA6330T chip (0x350,0x360,0x370,0x380)
irq - IRQ # for InterWave chip (3,5,9,11,12,15)
dma1 - DMA # for InterWave chip (0,1,3,5,6,7)
dma2 - DMA # for InterWave chip (0,1,3,5,6,7,-1=disable)
joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V)
midi - 1 = MIDI UART enable, 0 = MIDI UART disable (default)
pcm_voices - reserved PCM voices for the synthesizer (default 2)
effect - 1 = InterWave effects enable (default 0);
requires 8 voices
isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
with isapnp=0, the following options are available:
port - port # for InterWave chip (0x210,0x220,0x230,0x240,0x250,0x260)
port_tc - tone control (i2c bus) port # for TEA6330T chip (0x350,0x360,0x370,0x380)
irq - IRQ # for InterWave chip (3,5,9,11,12,15)
dma1 - DMA # for InterWave chip (0,1,3,5,6,7)
dma2 - DMA # for InterWave chip (0,1,3,5,6,7,-1=disable)
This module supports multiple cards, autoprobe and ISA PnP.
@ -1473,6 +1499,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
Module for Yamaha OPL3-SA2/SA3 sound cards.
isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
with isapnp=0, the following options are available:
port - control port # for OPL3-SA chip (0x370)
sb_port - SB port # for OPL3-SA chip (0x220,0x240)
wss_port - WSS port # for OPL3-SA chip (0x530,0xe80,0xf40,0x604)
@ -1481,7 +1511,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
irq - IRQ # for OPL3-SA chip (5,7,9,10)
dma1 - first DMA # for Yamaha OPL3-SA chip (0,1,3)
dma2 - second DMA # for Yamaha OPL3-SA chip (0,1,3), -1 = disable
isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
This module supports multiple cards and ISA PnP. It does not support
autoprobe (if ISA PnP is not used) thus all ports must be specified!!!
@ -1494,6 +1523,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
Module for sound cards based on OPTi 82c92x and Analog Devices AD1848 chips.
Module works with OAK Mozart cards as well.
isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
with isapnp=0, the following options are available:
port - port # for WSS chip (0x530,0xe80,0xf40,0x604)
mpu_port - port # for MPU-401 UART (0x300,0x310,0x320,0x330)
fm_port - port # for OPL3 device (0x388)
@ -1508,6 +1541,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
Module for sound cards based on OPTi 82c92x and Crystal CS4231 chips.
isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
with isapnp=0, the following options are available:
port - port # for WSS chip (0x530,0xe80,0xf40,0x604)
mpu_port - port # for MPU-401 UART (0x300,0x310,0x320,0x330)
fm_port - port # for OPL3 device (0x388)
@ -1523,6 +1560,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
Module for sound cards based on OPTi 82c93x chips.
isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
with isapnp=0, the following options are available:
port - port # for WSS chip (0x530,0xe80,0xf40,0x604)
mpu_port - port # for MPU-401 UART (0x300,0x310,0x320,0x330)
fm_port - port # for OPL3 device (0x388)
@ -1533,6 +1574,22 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
This module supports only one card, autoprobe and PnP.
Module snd-oxygen
-----------------
Module for sound cards based on the C-Media CMI8788 chip:
* Asound A-8788
* AuzenTech X-Meridian
* Bgears b-Enspirer
* Club3D Theatron DTS
* HT-Omega Claro
* Razer Barracuda AC-1
* Sondigo Inferno
This module supports autoprobe and multiple cards.
Power management is _not_ supported.
Module snd-pcxhr
----------------
@ -1647,6 +1704,12 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
SoundBlaster AWE 32 (PnP),
SoundBlaster AWE 64 PnP
mic_agc - Mic Auto-Gain-Control - 0 = disable, 1 = enable (default)
csp - ASP/CSP chip support - 0 = disable (default), 1 = enable
isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
with isapnp=0, the following options are available:
port - port # for SB DSP 4.x chip (0x220,0x240,0x260)
mpu_port - port # for MPU-401 UART (0x300,0x330), -1 = disable
awe_port - base port # for EMU8000 synthesizer (0x620,0x640,0x660)
@ -1654,9 +1717,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
irq - IRQ # for SB DSP 4.x chip (5,7,9,10)
dma8 - 8-bit DMA # for SB DSP 4.x chip (0,1,3)
dma16 - 16-bit DMA # for SB DSP 4.x chip (5,6,7)
mic_agc - Mic Auto-Gain-Control - 0 = disable, 1 = enable (default)
csp - ASP/CSP chip support - 0 = disable (default), 1 = enable
isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
This module supports multiple cards, autoprobe and ISA PnP.
@ -1739,18 +1799,21 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
Module for Turtle Beach Maui, Tropez and Tropez+ sound cards.
use_cs4232_midi - Use CS4232 MPU-401 interface
(inaccessibly located inside your computer)
isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
with isapnp=0, the following options are available:
cs4232_pcm_port - Port # for CS4232 PCM interface.
cs4232_pcm_irq - IRQ # for CS4232 PCM interface (5,7,9,11,12,15).
cs4232_mpu_port - Port # for CS4232 MPU-401 interface.
cs4232_mpu_irq - IRQ # for CS4232 MPU-401 interface (9,11,12,15).
use_cs4232_midi - Use CS4232 MPU-401 interface
(inaccessibly located inside your computer)
ics2115_port - Port # for ICS2115
ics2115_irq - IRQ # for ICS2115
fm_port - FM OPL-3 Port #
dma1 - DMA1 # for CS4232 PCM interface.
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)
@ -1965,6 +2028,16 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
This module supports multiple cards.
Module snd-virtuoso
-------------------
Module for sound cards based on the Asus AV200 chip, i.e.,
Xonar D2 and Xonar D2X.
This module supports autoprobe and multiple cards.
Power management is _not_ supported.
Module snd-vx222
----------------
@ -2135,6 +2208,23 @@ alias sound-slot-1 snd-ens1371
In this example, the interwave card is always loaded as the first card
(index 0) and ens1371 as the second (index 1).
Alternative (and new) way to fixate the slot assignment is to use
"slots" option of snd module. In the case above, specify like the
following:
options snd slots=snd-interwave,snd-ens1371
Then, the first slot (#0) is reserved for snd-interwave driver, and
the second (#1) for snd-ens1371. You can omit index option in each
driver if slots option is used (although you can still have them at
the same time as long as they don't conflict).
The slots option is especially useful for avoiding the possible
hot-plugging and the resultant slot conflict. For example, in the
case above again, the first two slots are already reserved. If any
other driver (e.g. snd-usb-audio) is loaded before snd-interwave or
snd-ens1371, it will be assigned to the third or later slot.
ALSA PCM devices to OSS devices mapping
=======================================

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Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
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6
Documentation/sound/alsa/soc/DAI.txt
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@ -1,5 +1,5 @@
ASoC currently supports the three main Digital Audio Interfaces (DAI) found on
SoC controllers and portable audio CODECS today, namely AC97, I2S and PCM.
SoC controllers and portable audio CODECs today, namely AC97, I2S and PCM.
AC97
@ -25,7 +25,7 @@ left/right clock (LRC) synchronise the link. I2S is flexible in that either the
controller or CODEC can drive (master) the BCLK and LRC clock lines. Bit clock
usually varies depending on the sample rate and the master system clock
(SYSCLK). LRCLK is the same as the sample rate. A few devices support separate
ADC and DAC LRCLK's, this allows for simultaneous capture and playback at
ADC and DAC LRCLKs, this allows for simultaneous capture and playback at
different sample rates.
I2S has several different operating modes:-
@ -35,7 +35,7 @@ I2S has several different operating modes:-
o Left Justified - MSB is transmitted on transition of LRC.
o Right Justified - MSB is transmitted sample size BCLK's before LRC
o Right Justified - MSB is transmitted sample size BCLKs before LRC
transition.
PCM

10
Documentation/sound/alsa/soc/clocking.txt
View File

@ -13,7 +13,7 @@ or SYSCLK). This audio master clock can be derived from a number of sources
(e.g. crystal, PLL, CPU clock) and is responsible for producing the correct
audio playback and capture sample rates.
Some master clocks (e.g. PLL's and CPU based clocks) are configurable in that
Some master clocks (e.g. PLLs and CPU based clocks) are configurable in that
their speed can be altered by software (depending on the system use and to save
power). Other master clocks are fixed at a set frequency (i.e. crystals).
@ -41,11 +41,11 @@ BCLK = LRC * x
BCLK = LRC * Channels * Word Size
This relationship depends on the codec or SoC CPU in particular. In general
it's best to configure BCLK to the lowest possible speed (depending on your
rate, number of channels and wordsize) to save on power.
it is best to configure BCLK to the lowest possible speed (depending on your
rate, number of channels and word size) to save on power.
It's also desirable to use the codec (if possible) to drive (or master) the
audio clocks as it's usually gives more accurate sample rates than the CPU.
It is also desirable to use the codec (if possible) to drive (or master) the
audio clocks as it usually gives more accurate sample rates than the CPU.

53
Documentation/sound/alsa/soc/codec.txt
View File

@ -9,7 +9,7 @@ code should be added to the platform and machine drivers respectively.
Each codec driver *must* provide the following features:-
1) Codec DAI and PCM configuration
2) Codec control IO - using I2C, 3 Wire(SPI) or both API's
2) Codec control IO - using I2C, 3 Wire(SPI) or both APIs
3) Mixers and audio controls
4) Codec audio operations
@ -19,7 +19,7 @@ Optionally, codec drivers can also provide:-
6) DAPM event handler.
7) DAC Digital mute control.
It's probably best to use this guide in conjunction with the existing codec
Its probably best to use this guide in conjunction with the existing codec
driver code in sound/soc/codecs/
ASoC Codec driver breakdown
@ -27,8 +27,8 @@ ASoC Codec driver breakdown
1 - Codec DAI and PCM configuration
-----------------------------------
Each codec driver must have a struct snd_soc_codec_dai to define it's DAI and
PCM's capabilities and operations. This struct is exported so that it can be
Each codec driver must have a struct snd_soc_codec_dai to define its DAI and
PCM capabilities and operations. This struct is exported so that it can be
registered with the core by your machine driver.
e.g.
@ -67,18 +67,18 @@ EXPORT_SYMBOL_GPL(wm8731_dai);
2 - Codec control IO
--------------------
The codec can usually be controlled via an I2C or SPI style interface (AC97
combines control with data in the DAI). The codec drivers will have to provide
functions to read and write the codec registers along with supplying a register
cache:-
The codec can usually be controlled via an I2C or SPI style interface
(AC97 combines control with data in the DAI). The codec drivers provide
functions to read and write the codec registers along with supplying a
register cache:-
/* IO control data and register cache */
void *control_data; /* codec control (i2c/3wire) data */
void *reg_cache;
void *control_data; /* codec control (i2c/3wire) data */
void *reg_cache;
Codec read/write should do any data formatting and call the hardware read write
below to perform the IO. These functions are called by the core and alsa when
performing DAPM or changing the mixer:-
Codec read/write should do any data formatting and call the hardware
read write below to perform the IO. These functions are called by the
core and ALSA when performing DAPM or changing the mixer:-
unsigned int (*read)(struct snd_soc_codec *, unsigned int);
int (*write)(struct snd_soc_codec *, unsigned int, unsigned int);
@ -131,7 +131,7 @@ Defines a stereo enumerated control
4 - Codec Audio Operations
--------------------------
The codec driver also supports the following alsa operations:-
The codec driver also supports the following ALSA operations:-
/* SoC audio ops */
struct snd_soc_ops {
@ -142,15 +142,15 @@ struct snd_soc_ops {
int (*prepare)(struct snd_pcm_substream *);
};
Please refer to the alsa driver PCM documentation for details.
Please refer to the ALSA driver PCM documentation for details.
http://www.alsa-project.org/~iwai/writing-an-alsa-driver/c436.htm
5 - DAPM description.
---------------------
The Dynamic Audio Power Management description describes the codec's power
components, their relationships and registers to the ASoC core. Please read
dapm.txt for details of building the description.
The Dynamic Audio Power Management description describes the codec power
components and their relationships and registers to the ASoC core.
Please read dapm.txt for details of building the description.
Please also see the examples in other codec drivers.
@ -158,8 +158,8 @@ Please also see the examples in other codec drivers.
6 - DAPM event handler
----------------------
This function is a callback that handles codec domain PM calls and system
domain PM calls (e.g. suspend and resume). It's used to put the codec to sleep
when not in use.
domain PM calls (e.g. suspend and resume). It is used to put the codec
to sleep when not in use.
Power states:-
@ -175,13 +175,14 @@ Power states:-
SNDRV_CTL_POWER_D3cold: /* Everything Off, without power */
7 - Codec DAC digital mute control.
------------------------------------
Most codecs have a digital mute before the DAC's that can be used to minimise
any system noise. The mute stops any digital data from entering the DAC.
7 - Codec DAC digital mute control
----------------------------------
Most codecs have a digital mute before the DACs that can be used to
minimise any system noise. The mute stops any digital data from
entering the DAC.
A callback can be created that is called by the core for each codec DAI when the
mute is applied or freed.
A callback can be created that is called by the core for each codec DAI
when the mute is applied or freed.
i.e.

51
Documentation/sound/alsa/soc/dapm.txt
View File

@ -4,20 +4,20 @@ Dynamic Audio Power Management for Portable Devices
1. Description
==============
Dynamic Audio Power Management (DAPM) is designed to allow portable Linux devices
to use the minimum amount of power within the audio subsystem at all times. It
is independent of other kernel PM and as such, can easily co-exist with the
other PM systems.
Dynamic Audio Power Management (DAPM) is designed to allow portable
Linux devices to use the minimum amount of power within the audio
subsystem at all times. It is independent of other kernel PM and as
such, can easily co-exist with the other PM systems.
DAPM is also completely transparent to all user space applications as all power
switching is done within the ASoC core. No code changes or recompiling are
required for user space applications. DAPM makes power switching decisions based
upon any audio stream (capture/playback) activity and audio mixer settings
within the device.
DAPM is also completely transparent to all user space applications as
all power switching is done within the ASoC core. No code changes or
recompiling are required for user space applications. DAPM makes power
switching decisions based upon any audio stream (capture/playback)
activity and audio mixer settings within the device.
DAPM spans the whole machine. It covers power control within the entire audio
subsystem, this includes internal codec power blocks and machine level power
systems.
DAPM spans the whole machine. It covers power control within the entire
audio subsystem, this includes internal codec power blocks and machine
level power systems.
There are 4 power domains within DAPM
@ -34,7 +34,7 @@ There are 4 power domains within DAPM
Automatically set when mixer and mux settings are changed by the user.
e.g. alsamixer, amixer.
4. Stream domain - DAC's and ADC's.
4. Stream domain - DACs and ADCs.
Enabled and disabled when stream playback/capture is started and
stopped respectively. e.g. aplay, arecord.
@ -51,7 +51,7 @@ widgets hereafter.
Audio DAPM widgets fall into a number of types:-
o Mixer - Mixes several analog signals into a single analog signal.
o Mux - An analog switch that outputs only 1 of it's inputs.
o Mux - An analog switch that outputs only one of many inputs.
o PGA - A programmable gain amplifier or attenuation widget.
o ADC - Analog to Digital Converter
o DAC - Digital to Analog Converter
@ -78,14 +78,14 @@ parameters for stream name and kcontrols.
2.1 Stream Domain Widgets
-------------------------
Stream Widgets relate to the stream power domain and only consist of ADC's
(analog to digital converters) and DAC's (digital to analog converters).
Stream Widgets relate to the stream power domain and only consist of ADCs
(analog to digital converters) and DACs (digital to analog converters).
Stream widgets have the following format:-
SND_SOC_DAPM_DAC(name, stream name, reg, shift, invert),
NOTE: the stream name must match the corresponding stream name in your codecs
NOTE: the stream name must match the corresponding stream name in your codec
snd_soc_codec_dai.
e.g. stream widgets for HiFi playback and capture
@ -97,7 +97,7 @@ SND_SOC_DAPM_ADC("HiFi ADC", "HiFi Capture", REG, 2, 1),
2.2 Path Domain Widgets
-----------------------
Path domain widgets have a ability to control or effect the audio signal or
Path domain widgets have a ability to control or affect the audio signal or
audio paths within the audio subsystem. They have the following form:-
SND_SOC_DAPM_PGA(name, reg, shift, invert, controls, num_controls)
@ -149,7 +149,7 @@ SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias),
2.4 Codec Domain
----------------
The Codec power domain has no widgets and is handled by the codecs DAPM event
The codec power domain has no widgets and is handled by the codecs DAPM event
handler. This handler is called when the codec powerstate is changed wrt to any
stream event or by kernel PM events.
@ -158,8 +158,8 @@ stream event or by kernel PM events.
-------------------
Sometimes widgets exist in the codec or machine audio map that don't have any
corresponding register bit for power control. In this case it's necessary to
create a virtual widget - a widget with no control bits e.g.
corresponding soft power control. In this case it is necessary to create
a virtual widget - a widget with no control bits e.g.
SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_DAPM_NOPM, 0, 0, NULL, 0),
@ -172,13 +172,14 @@ subsystem individually with a call to snd_soc_dapm_new_control().
3. Codec Widget Interconnections
================================
Widgets are connected to each other within the codec and machine by audio
paths (called interconnections). Each interconnection must be defined in order
to create a map of all audio paths between widgets.
Widgets are connected to each other within the codec and machine by audio paths
(called interconnections). Each interconnection must be defined in order to
create a map of all audio paths between widgets.
This is easiest with a diagram of the codec (and schematic of the machine audio
system), as it requires joining widgets together via their audio signal paths.
i.e. from the WM8731 codec's output mixer (wm8731.c)
e.g., from the WM8731 output mixer (wm8731.c)
The WM8731 output mixer has 3 inputs (sources)

12
Documentation/sound/alsa/soc/machine.txt
View File

@ -16,7 +16,7 @@ struct snd_soc_machine {
int (*remove)(struct platform_device *pdev);
/* the pre and post PM functions are used to do any PM work before and
* after the codec and DAI's do any PM work. */
* after the codec and DAIs do any PM work. */
int (*suspend_pre)(struct platform_device *pdev, pm_message_t state);
int (*suspend_post)(struct platform_device *pdev, pm_message_t state);
int (*resume_pre)(struct platform_device *pdev);
@ -38,7 +38,7 @@ probe/remove are optional. Do any machine specific probe here.
suspend()/resume()
------------------
The machine driver has pre and post versions of suspend and resume to take care
of any machine audio tasks that have to be done before or after the codec, DAI's
of any machine audio tasks that have to be done before or after the codec, DAIs
and DMA is suspended and resumed. Optional.
@ -49,10 +49,10 @@ The machine specific audio operations can be set here. Again this is optional.
Machine DAI Configuration
-------------------------
The machine DAI configuration glues all the codec and CPU DAI's together. It can
The machine DAI configuration glues all the codec and CPU DAIs together. It can
also be used to set up the DAI system clock and for any machine related DAI
initialisation e.g. the machine audio map can be connected to the codec audio
map, unconnnected codec pins can be set as such. Please see corgi.c, spitz.c
map, unconnected codec pins can be set as such. Please see corgi.c, spitz.c
for examples.
struct snd_soc_dai_link is used to set up each DAI in your machine. e.g.
@ -67,7 +67,7 @@ static struct snd_soc_dai_link corgi_dai = {
.ops = &corgi_ops,
};
struct snd_soc_machine then sets up the machine with it's DAI's. e.g.
struct snd_soc_machine then sets up the machine with it's DAIs. e.g.
/* corgi audio machine driver */
static struct snd_soc_machine snd_soc_machine_corgi = {
@ -110,4 +110,4 @@ details.
Machine Controls
----------------
Machine specific audio mixer controls can be added in the dai init function.
Machine specific audio mixer controls can be added in the DAI init function.

42
Documentation/sound/alsa/soc/overview.txt
View File

@ -1,25 +1,26 @@
ALSA SoC Layer
==============
The overall project goal of the ALSA System on Chip (ASoC) layer is to provide
better ALSA support for embedded system-on-chip processors (e.g. pxa2xx, au1x00,
iMX, etc) and portable audio codecs. Currently there is some support in the
kernel for SoC audio, however it has some limitations:-
The overall project goal of the ALSA System on Chip (ASoC) layer is to
provide better ALSA support for embedded system-on-chip processors (e.g.
pxa2xx, au1x00, iMX, etc) and portable audio codecs. Prior to the ASoC
subsystem there was some support in the kernel for SoC audio, however it
had some limitations:-
* Currently, codec drivers are often tightly coupled to the underlying SoC
CPU. This is not ideal and leads to code duplication i.e. Linux now has 4
different wm8731 drivers for 4 different SoC platforms.
* Codec drivers were often tightly coupled to the underlying SoC
CPU. This is not ideal and leads to code duplication - for example,
Linux had different wm8731 drivers for 4 different SoC platforms.
* There is no standard method to signal user initiated audio events (e.g.
* There was no standard method to signal user initiated audio events (e.g.
Headphone/Mic insertion, Headphone/Mic detection after an insertion
event). These are quite common events on portable devices and often require
machine specific code to re-route audio, enable amps, etc., after such an
event.
* Current drivers tend to power up the entire codec when playing
(or recording) audio. This is fine for a PC, but tends to waste a lot of
power on portable devices. There is also no support for saving power via
changing codec oversampling rates, bias currents, etc.
* Drivers tended to power up the entire codec when playing (or
recording) audio. This is fine for a PC, but tends to waste a lot of
power on portable devices. There was also no support for saving
power via changing codec oversampling rates, bias currents, etc.
ASoC Design
@ -31,12 +32,13 @@ features :-
* Codec independence. Allows reuse of codec drivers on other platforms
and machines.
* Easy I2S/PCM audio interface setup between codec and SoC. Each SoC interface
and codec registers it's audio interface capabilities with the core and are
subsequently matched and configured when the application hw params are known.
* Easy I2S/PCM audio interface setup between codec and SoC. Each SoC
interface and codec registers it's audio interface capabilities with the
core and are subsequently matched and configured when the application
hardware parameters are known.
* Dynamic Audio Power Management (DAPM). DAPM automatically sets the codec to
it's minimum power state at all times. This includes powering up/down
its minimum power state at all times. This includes powering up/down
internal power blocks depending on the internal codec audio routing and any
active streams.
@ -45,16 +47,16 @@ features :-
signals the codec when to change power states.
* Machine specific controls: Allow machines to add controls to the sound card
(e.g. volume control for speaker amp).
(e.g. volume control for speaker amplifier).
To achieve all this, ASoC basically splits an embedded audio system into 3
components :-
* Codec driver: The codec driver is platform independent and contains audio
controls, audio interface capabilities, codec dapm definition and codec IO
controls, audio interface capabilities, codec DAPM definition and codec IO
functions.
* Platform driver: The platform driver contains the audio dma engine and audio
* Platform driver: The platform driver contains the audio DMA engine and audio
interface drivers (e.g. I2S, AC97, PCM) for that platform.
* Machine driver: The machine driver handles any machine specific controls and
@ -81,4 +83,4 @@ machine.txt: Machine driver internals.
pop_clicks.txt: How to minimise audio artifacts.
clocking.txt: ASoC clocking for best power performance.
clocking.txt: ASoC clocking for best power performance.

6
Documentation/sound/alsa/soc/platform.txt
View File

@ -8,7 +8,7 @@ specific code.
Audio DMA
=========
The platform DMA driver optionally supports the following alsa operations:-
The platform DMA driver optionally supports the following ALSA operations:-
/* SoC audio ops */
struct snd_soc_ops {
@ -38,7 +38,7 @@ struct snd_soc_platform {
struct snd_pcm_ops *pcm_ops;
};
Please refer to the alsa driver documentation for details of audio DMA.
Please refer to the ALSA driver documentation for details of audio DMA.
http://www.alsa-project.org/~iwai/writing-an-alsa-driver/c436.htm
An example DMA driver is soc/pxa/pxa2xx-pcm.c
@ -52,7 +52,7 @@ Each SoC DAI driver must provide the following features:-
1) Digital audio interface (DAI) description
2) Digital audio interface configuration
3) PCM's description
4) Sysclk configuration
4) SYSCLK configuration
5) Suspend and resume (optional)
Please see codec.txt for a description of items 1 - 4.

10
Documentation/sound/alsa/soc/pops_clicks.txt
View File

@ -15,11 +15,11 @@ click every time a component power state is changed.
Minimising Playback Pops and Clicks
===================================
Playback pops in portable audio subsystems cannot be completely eliminated atm,
however future audio codec hardware will have better pop and click suppression.
Pops can be reduced within playback by powering the audio components in a
specific order. This order is different for startup and shutdown and follows
some basic rules:-
Playback pops in portable audio subsystems cannot be completely eliminated
currently, however future audio codec hardware will have better pop and click
suppression. Pops can be reduced within playback by powering the audio
components in a specific order. This order is different for startup and
shutdown and follows some basic rules:-
Startup Order :- DAC --> Mixers --> Output PGA --> Digital Unmute

3
MAINTAINERS
View File

@ -3571,6 +3571,9 @@ S: Maintained
SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT
P: Liam Girdwood
M: liam.girdwood@wolfsonmicro.com
P: Mark Brown
M: broonie@opensource.wolfsonmicro.com
T: git opensource.wolfsonmicro.com/linux-2.6-asoc
L: alsa-devel@alsa-project.org (subscribers-only)
S: Supported

1
drivers/input/touchscreen/ucb1400_ts.c
View File

@ -27,7 +27,6 @@
#include <linux/kthread.h>
#include <linux/freezer.h>
#include <sound/driver.h>
#include <sound/core.h>
#include <sound/ac97_codec.h>

1
drivers/media/video/cx88/cx88-alsa.c
View File

@ -33,7 +33,6 @@
#include <linux/pci.h>
#include <asm/delay.h>
#include <sound/driver.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>

1
drivers/media/video/saa7134/saa7134-alsa.c
View File

@ -21,7 +21,6 @@
#include <linux/time.h>
#include <linux/wait.h>
#include <linux/module.h>
#include <sound/driver.h>
#include <sound/core.h>
#include <sound/control.h>
#include <sound/pcm.h>

1
drivers/media/video/saa7134/saa7134.h
View File

@ -38,7 +38,6 @@
#include <media/ir-common.h>
#include <media/ir-kbd-i2c.h>
#include <media/videobuf-dma-sg.h>
#include <sound/driver.h>
#include <sound/core.h>
#include <sound/pcm.h>
#if defined(CONFIG_VIDEO_SAA7134_DVB) || defined(CONFIG_VIDEO_SAA7134_DVB_MODULE)

1
drivers/usb/gadget/gmidi.c
View File

@ -24,7 +24,6 @@
#include <linux/utsname.h>
#include <linux/device.h>
#include <sound/driver.h>
#include <sound/core.h>
#include <sound/initval.h>
#include <sound/rawmidi.h>

1
include/asm-arm/arch-omap/eac.h
View File

@ -31,7 +31,6 @@
#include <asm/arch/hardware.h>
#include <asm/irq.h>
#include <sound/driver.h>
#include <sound/core.h>
/* master codec clock source */

1
include/asm-arm/arch-omap/omap-alsa.h
View File

@ -40,7 +40,6 @@
#ifndef __OMAP_ALSA_H
#define __OMAP_ALSA_H
#include <sound/driver.h>
#include <asm/arch/dma.h>
#include <sound/core.h>
#include <sound/pcm.h>

1
include/asm-arm/arch-pxa/audio.h
View File

@ -1,7 +1,6 @@
#ifndef __ASM_ARCH_AUDIO_H__
#define __ASM_ARCH_AUDIO_H__
#include <sound/driver.h>
#include <sound/core.h>
#include <sound/pcm.h>

2
include/sound/ad1848.h
View File

@ -48,7 +48,7 @@
#define AD1848_IFACE_CTRL 0x09 /* interface control - bits 7-2 MCE */
#define AD1848_PIN_CTRL 0x0a /* pin control */
#define AD1848_TEST_INIT 0x0b /* test and initialization */
#define AD1848_MISC_INFO 0x0c /* miscellaneaous information */
#define AD1848_MISC_INFO 0x0c /* miscellaneous information */
#define AD1848_LOOPBACK 0x0d /* loopback control */
#define AD1848_DATA_UPR_CNT 0x0e /* playback/capture upper base count */
#define AD1848_DATA_LWR_CNT 0x0f /* playback/capture lower base count */

134
include/sound/ainstr_fm.h
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@ -1,134 +0,0 @@
/*
* Advanced Linux Sound Architecture
*
* FM (OPL2/3) Instrument Format
* Copyright (c) 2000 Uros Bizjak <uros@kss-loka.si>
*
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
*/
#ifndef __SOUND_AINSTR_FM_H
#define __SOUND_AINSTR_FM_H
#ifndef __KERNEL__
#include <asm/types.h>
#include <asm/byteorder.h>
#endif
/*
* share types (share ID 1)
*/
#define FM_SHARE_FILE 0
/*
* FM operator
*/
struct fm_operator {
unsigned char am_vib;
unsigned char ksl_level;
unsigned char attack_decay;
unsigned char sustain_release;
unsigned char wave_select;
};
/*
* Instrument
*/
#define FM_PATCH_OPL2 0x01 /* OPL2 2 operators FM instrument */
#define FM_PATCH_OPL3 0x02 /* OPL3 4 operators FM instrument */
struct fm_instrument {
unsigned int share_id[4]; /* share id - zero = no sharing */
unsigned char type; /* instrument type */
struct fm_operator op[4];
unsigned char feedback_connection[2];
unsigned char echo_delay;
unsigned char echo_atten;
unsigned char chorus_spread;
unsigned char trnsps;
unsigned char fix_dur;
unsigned char modes;
unsigned char fix_key;
};
/*
*
* Kernel <-> user space
* Hardware (CPU) independent section
*
* * = zero or more
* + = one or more
*
* fm_xinstrument FM_STRU_INSTR
*
*/
#define FM_STRU_INSTR __cpu_to_be32(('I'<<24)|('N'<<16)|('S'<<8)|'T')
/*
* FM operator
*/
struct fm_xoperator {
__u8 am_vib;
__u8 ksl_level;
__u8 attack_decay;
__u8 sustain_release;
__u8 wave_select;
};
/*
* Instrument
*/
struct fm_xinstrument {
__u32 stype; /* structure type */
__u32 share_id[4]; /* share id - zero = no sharing */
__u8 type; /* instrument type */
struct fm_xoperator op[4]; /* fm operators */
__u8 feedback_connection[2];
__u8 echo_delay;
__u8 echo_atten;
__u8 chorus_spread;
__u8 trnsps;
__u8 fix_dur;
__u8 modes;
__u8 fix_key;
};
#ifdef __KERNEL__
#include "seq_instr.h"
int snd_seq_fm_init(struct snd_seq_kinstr_ops * ops,
struct snd_seq_kinstr_ops * next);
#endif
/* typedefs for compatibility to user-space */
typedef struct fm_xoperator fm_xoperator_t;
typedef struct fm_xinstrument fm_xinstrument_t;
#endif /* __SOUND_AINSTR_FM_H */

229
include/sound/ainstr_gf1.h
View File

@ -1,229 +0,0 @@
/*
* Advanced Linux Sound Architecture
*
* GF1 (GUS) Patch Instrument Format
* Copyright (c) 1994-99 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
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
*/
#ifndef __SOUND_AINSTR_GF1_H
#define __SOUND_AINSTR_GF1_H
#ifndef __KERNEL__
#include <asm/types.h>
#include <asm/byteorder.h>
#endif
/*
* share types (share ID 1)
*/
#define GF1_SHARE_FILE 0
/*
* wave formats
*/
#define GF1_WAVE_16BIT 0x0001 /* 16-bit wave */
#define GF1_WAVE_UNSIGNED 0x0002 /* unsigned wave */
#define GF1_WAVE_INVERT 0x0002 /* same as unsigned wave */
#define GF1_WAVE_BACKWARD 0x0004 /* backward mode (maybe used for reverb or ping-ping loop) */
#define GF1_WAVE_LOOP 0x0008 /* loop mode */
#define GF1_WAVE_BIDIR 0x0010 /* bidirectional mode */
#define GF1_WAVE_STEREO 0x0100 /* stereo mode */
#define GF1_WAVE_ULAW 0x0200 /* uLaw compression mode */
/*
* Wavetable definitions
*/
struct gf1_wave {
unsigned int share_id[4]; /* share id - zero = no sharing */
unsigned int format; /* wave format */
struct {
unsigned int number; /* some other ID for this instrument */
unsigned int memory; /* begin of waveform in onboard memory */
unsigned char *ptr; /* pointer to waveform in system memory */
} address;
unsigned int size; /* size of waveform in samples */
unsigned int start; /* start offset in samples * 16 (lowest 4 bits - fraction) */
unsigned int loop_start; /* bits loop start offset in samples * 16 (lowest 4 bits - fraction) */
unsigned int loop_end; /* loop start offset in samples * 16 (lowest 4 bits - fraction) */
unsigned short loop_repeat; /* loop repeat - 0 = forever */
unsigned char flags; /* GF1 patch flags */
unsigned char pad;
unsigned int sample_rate; /* sample rate in Hz */
unsigned int low_frequency; /* low frequency range */
unsigned int high_frequency; /* high frequency range */
unsigned int root_frequency; /* root frequency range */
signed short tune;
unsigned char balance;
unsigned char envelope_rate[6];
unsigned char envelope_offset[6];
unsigned char tremolo_sweep;
unsigned char tremolo_rate;
unsigned char tremolo_depth;
unsigned char vibrato_sweep;
unsigned char vibrato_rate;
unsigned char vibrato_depth;
unsigned short scale_frequency;
unsigned short scale_factor; /* 0-2048 or 0-2 */
struct gf1_wave *next;
};
/*
* Instrument
*/
#define IWFFFF_EXCLUDE_NONE 0x0000 /* exclusion mode - none */
#define IWFFFF_EXCLUDE_SINGLE 0x0001 /* exclude single - single note from the instrument group */
#define IWFFFF_EXCLUDE_MULTIPLE 0x0002 /* exclude multiple - stop only same note from this instrument */
#define IWFFFF_EFFECT_NONE 0
#define IWFFFF_EFFECT_REVERB 1
#define IWFFFF_EFFECT_CHORUS 2
#define IWFFFF_EFFECT_ECHO 3
struct gf1_instrument {
unsigned short exclusion;
unsigned short exclusion_group; /* 0 - none, 1-65535 */
unsigned char effect1; /* effect 1 */
unsigned char effect1_depth; /* 0-127 */
unsigned char effect2; /* effect 2 */
unsigned char effect2_depth; /* 0-127 */
struct gf1_wave *wave; /* first waveform */
};
/*
*
* Kernel <-> user space
* Hardware (CPU) independent section
*
* * = zero or more
* + = one or more
*
* gf1_xinstrument IWFFFF_STRU_INSTR
* +gf1_xwave IWFFFF_STRU_WAVE
*
*/
#define GF1_STRU_WAVE __cpu_to_be32(('W'<<24)|('A'<<16)|('V'<<8)|'E')
#define GF1_STRU_INSTR __cpu_to_be32(('I'<<24)|('N'<<16)|('S'<<8)|'T')
/*
* Wavetable definitions
*/
struct gf1_xwave {
__u32 stype; /* structure type */
__u32 share_id[4]; /* share id - zero = no sharing */
__u32 format; /* wave format */
__u32 size; /* size of waveform in samples */
__u32 start; /* start offset in samples * 16 (lowest 4 bits - fraction) */
__u32 loop_start; /* bits loop start offset in samples * 16 (lowest 4 bits - fraction) */
__u32 loop_end; /* loop start offset in samples * 16 (lowest 4 bits - fraction) */
__u16 loop_repeat; /* loop repeat - 0 = forever */
__u8 flags; /* GF1 patch flags */
__u8 pad;
__u32 sample_rate; /* sample rate in Hz */
__u32 low_frequency; /* low frequency range */
__u32 high_frequency; /* high frequency range */
__u32 root_frequency; /* root frequency range */
__s16 tune;
__u8 balance;
__u8 envelope_rate[6];
__u8 envelope_offset[6];
__u8 tremolo_sweep;
__u8 tremolo_rate;
__u8 tremolo_depth;
__u8 vibrato_sweep;
__u8 vibrato_rate;
__u8 vibrato_depth;
__u16 scale_frequency;
__u16 scale_factor; /* 0-2048 or 0-2 */
};
/*
* Instrument
*/
struct gf1_xinstrument {
__u32 stype;
__u16 exclusion;
__u16 exclusion_group; /* 0 - none, 1-65535 */
__u8 effect1; /* effect 1 */
__u8 effect1_depth; /* 0-127 */
__u8 effect2; /* effect 2 */
__u8 effect2_depth; /* 0-127 */
};
/*
* Instrument info
*/
#define GF1_INFO_ENVELOPE (1<<0)
#define GF1_INFO_TREMOLO (1<<1)
#define GF1_INFO_VIBRATO (1<<2)
struct gf1_info {
unsigned char flags; /* supported wave flags */
unsigned char pad[3];
unsigned int features; /* supported features */
unsigned int max8_len; /* maximum 8-bit wave length */
unsigned int max16_len; /* maximum 16-bit wave length */
};
#ifdef __KERNEL__
#include "seq_instr.h"
struct snd_gf1_ops {
void *private_data;
int (*info)(void *private_data, struct gf1_info *info);
int (*put_sample)(void *private_data, struct gf1_wave *wave,
char __user *data, long len, int atomic);
int (*get_sample)(void *private_data, struct gf1_wave *wave,
char __user *data, long len, int atomic);
int (*remove_sample)(void *private_data, struct gf1_wave *wave,
int atomic);
void (*notify)(void *private_data, struct snd_seq_kinstr *instr, int what);
struct snd_seq_kinstr_ops kops;
};
int snd_seq_gf1_init(struct snd_gf1_ops *ops,
void *private_data,
struct snd_seq_kinstr_ops *next);
#endif
/* typedefs for compatibility to user-space */
typedef struct gf1_xwave gf1_xwave_t;
typedef struct gf1_xinstrument gf1_xinstrument_t;
#endif /* __SOUND_AINSTR_GF1_H */

384
include/sound/ainstr_iw.h
View File

@ -1,384 +0,0 @@
/*
* Advanced Linux Sound Architecture
*
* InterWave FFFF Instrument Format
* Copyright (c) 1994-99 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
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
*/
#ifndef __SOUND_AINSTR_IW_H
#define __SOUND_AINSTR_IW_H
#ifndef __KERNEL__
#include <asm/types.h>
#include <asm/byteorder.h>
#endif
/*
* share types (share ID 1)
*/
#define IWFFFF_SHARE_FILE 0
/*
* wave formats
*/
#define IWFFFF_WAVE_16BIT 0x0001 /* 16-bit wave */
#define IWFFFF_WAVE_UNSIGNED 0x0002 /* unsigned wave */
#define IWFFFF_WAVE_INVERT 0x0002 /* same as unsigned wave */
#define IWFFFF_WAVE_BACKWARD 0x0004 /* backward mode (maybe used for reverb or ping-ping loop) */
#define IWFFFF_WAVE_LOOP 0x0008 /* loop mode */
#define IWFFFF_WAVE_BIDIR 0x0010 /* bidirectional mode */
#define IWFFFF_WAVE_ULAW 0x0020 /* uLaw compressed wave */
#define IWFFFF_WAVE_RAM 0x0040 /* wave is _preloaded_ in RAM (it is used for ROM simulation) */
#define IWFFFF_WAVE_ROM 0x0080 /* wave is in ROM */
#define IWFFFF_WAVE_STEREO 0x0100 /* wave is stereo */
/*
* Wavetable definitions
*/
struct iwffff_wave {
unsigned int share_id[4]; /* share id - zero = no sharing */
unsigned int format; /* wave format */
struct {
unsigned int number; /* some other ID for this wave */
unsigned int memory; /* begin of waveform in onboard memory */
unsigned char *ptr; /* pointer to waveform in system memory */
} address;
unsigned int size; /* size of waveform in samples */
unsigned int start; /* start offset in samples * 16 (lowest 4 bits - fraction) */
unsigned int loop_start; /* bits loop start offset in samples * 16 (lowest 4 bits - fraction) */
unsigned int loop_end; /* loop start offset in samples * 16 (lowest 4 bits - fraction) */
unsigned short loop_repeat; /* loop repeat - 0 = forever */
unsigned int sample_ratio; /* sample ratio (44100 * 1024 / rate) */
unsigned char attenuation; /* 0 - 127 (no corresponding midi controller) */
unsigned char low_note; /* lower frequency range for this waveform */
unsigned char high_note; /* higher frequency range for this waveform */
unsigned char pad;
struct iwffff_wave *next;
};
/*
* Layer
*/
#define IWFFFF_LFO_SHAPE_TRIANGLE 0
#define IWFFFF_LFO_SHAPE_POSTRIANGLE 1
struct iwffff_lfo {
unsigned short freq; /* (0-2047) 0.01Hz - 21.5Hz */
signed short depth; /* volume +- (0-255) 0.48675dB/step */
signed short sweep; /* 0 - 950 deciseconds */
unsigned char shape; /* see to IWFFFF_LFO_SHAPE_XXXX */
unsigned char delay; /* 0 - 255 deciseconds */
};
#define IWFFFF_ENV_FLAG_RETRIGGER 0x0001 /* flag - retrigger */
#define IWFFFF_ENV_MODE_ONE_SHOT 0x0001 /* mode - one shot */
#define IWFFFF_ENV_MODE_SUSTAIN 0x0002 /* mode - sustain */
#define IWFFFF_ENV_MODE_NO_SUSTAIN 0x0003 /* mode - no sustain */
#define IWFFFF_ENV_INDEX_VELOCITY 0x0001 /* index - velocity */
#define IWFFFF_ENV_INDEX_FREQUENCY 0x0002 /* index - frequency */
struct iwffff_env_point {
unsigned short offset;
unsigned short rate;
};
struct iwffff_env_record {
unsigned short nattack;
unsigned short nrelease;
unsigned short sustain_offset;
unsigned short sustain_rate;
unsigned short release_rate;
unsigned char hirange;
unsigned char pad;
struct iwffff_env_record *next;
/* points are stored here */
/* count of points = nattack + nrelease */
};
struct iwffff_env {
unsigned char flags;
unsigned char mode;
unsigned char index;
unsigned char pad;
struct iwffff_env_record *record;
};
#define IWFFFF_LAYER_FLAG_RETRIGGER 0x0001 /* retrigger */
#define IWFFFF_LAYER_VELOCITY_TIME 0x0000 /* velocity mode = time */
#define IWFFFF_LAYER_VELOCITY_RATE 0x0001 /* velocity mode = rate */
#define IWFFFF_LAYER_EVENT_KUP 0x0000 /* layer event - key up */
#define IWFFFF_LAYER_EVENT_KDOWN 0x0001 /* layer event - key down */
#define IWFFFF_LAYER_EVENT_RETRIG 0x0002 /* layer event - retrigger */
#define IWFFFF_LAYER_EVENT_LEGATO 0x0003 /* layer event - legato */
struct iwffff_layer {
unsigned char flags;
unsigned char velocity_mode;
unsigned char layer_event;
unsigned char low_range; /* range for layer based */
unsigned char high_range; /* on either velocity or frequency */
unsigned char pan; /* pan offset from CC1 (0 left - 127 right) */
unsigned char pan_freq_scale; /* position based on frequency (0-127) */
unsigned char attenuation; /* 0-127 (no corresponding midi controller) */
struct iwffff_lfo tremolo; /* tremolo effect */
struct iwffff_lfo vibrato; /* vibrato effect */
unsigned short freq_scale; /* 0-2048, 1024 is equal to semitone scaling */
unsigned char freq_center; /* center for keyboard frequency scaling */
unsigned char pad;
struct iwffff_env penv; /* pitch envelope */
struct iwffff_env venv; /* volume envelope */
struct iwffff_wave *wave;
struct iwffff_layer *next;
};
/*
* Instrument
*/
#define IWFFFF_EXCLUDE_NONE 0x0000 /* exclusion mode - none */
#define IWFFFF_EXCLUDE_SINGLE 0x0001 /* exclude single - single note from the instrument group */
#define IWFFFF_EXCLUDE_MULTIPLE 0x0002 /* exclude multiple - stop only same note from this instrument */
#define IWFFFF_LAYER_NONE 0x0000 /* not layered */
#define IWFFFF_LAYER_ON 0x0001 /* layered */
#define IWFFFF_LAYER_VELOCITY 0x0002 /* layered by velocity */
#define IWFFFF_LAYER_FREQUENCY 0x0003 /* layered by frequency */
#define IWFFFF_EFFECT_NONE 0
#define IWFFFF_EFFECT_REVERB 1
#define IWFFFF_EFFECT_CHORUS 2
#define IWFFFF_EFFECT_ECHO 3
struct iwffff_instrument {
unsigned short exclusion;
unsigned short layer_type;
unsigned short exclusion_group; /* 0 - none, 1-65535 */
unsigned char effect1; /* effect 1 */
unsigned char effect1_depth; /* 0-127 */
unsigned char effect2; /* effect 2 */
unsigned char effect2_depth; /* 0-127 */
struct iwffff_layer *layer; /* first layer */
};
/*
*
* Kernel <-> user space
* Hardware (CPU) independent section
*
* * = zero or more
* + = one or more
*
* iwffff_xinstrument IWFFFF_STRU_INSTR
* +iwffff_xlayer IWFFFF_STRU_LAYER
* *iwffff_xenv_record IWFFFF_STRU_ENV_RECT (tremolo)
* *iwffff_xenv_record IWFFFF_STRU_EVN_RECT (vibrato)
* +iwffff_xwave IWFFFF_STRU_WAVE
*
*/
#define IWFFFF_STRU_WAVE __cpu_to_be32(('W'<<24)|('A'<<16)|('V'<<8)|'E')
#define IWFFFF_STRU_ENV_RECP __cpu_to_be32(('E'<<24)|('N'<<16)|('R'<<8)|'P')
#define IWFFFF_STRU_ENV_RECV __cpu_to_be32(('E'<<24)|('N'<<16)|('R'<<8)|'V')
#define IWFFFF_STRU_LAYER __cpu_to_be32(('L'<<24)|('A'<<16)|('Y'<<8)|'R')
#define IWFFFF_STRU_INSTR __cpu_to_be32(('I'<<24)|('N'<<16)|('S'<<8)|'T')
/*
* Wavetable definitions
*/
struct iwffff_xwave {
__u32 stype; /* structure type */
__u32 share_id[4]; /* share id - zero = no sharing */
__u32 format; /* wave format */
__u32 offset; /* offset to ROM (address) */
__u32 size; /* size of waveform in samples */
__u32 start; /* start offset in samples * 16 (lowest 4 bits - fraction) */
__u32 loop_start; /* bits loop start offset in samples * 16 (lowest 4 bits - fraction) */
__u32 loop_end; /* loop start offset in samples * 16 (lowest 4 bits - fraction) */
__u16 loop_repeat; /* loop repeat - 0 = forever */
__u32 sample_ratio; /* sample ratio (44100 * 1024 / rate) */
__u8 attenuation; /* 0 - 127 (no corresponding midi controller) */
__u8 low_note; /* lower frequency range for this waveform */
__u8 high_note; /* higher frequency range for this waveform */
__u8 pad;
};
/*
* Layer
*/
struct iwffff_xlfo {
__u16 freq; /* (0-2047) 0.01Hz - 21.5Hz */
__s16 depth; /* volume +- (0-255) 0.48675dB/step */
__s16 sweep; /* 0 - 950 deciseconds */
__u8 shape; /* see to ULTRA_IW_LFO_SHAPE_XXXX */
__u8 delay; /* 0 - 255 deciseconds */
};
struct iwffff_xenv_point {
__u16 offset;
__u16 rate;
};
struct iwffff_xenv_record {
__u32 stype;
__u16 nattack;
__u16 nrelease;
__u16 sustain_offset;
__u16 sustain_rate;
__u16 release_rate;
__u8 hirange;
__u8 pad;
/* points are stored here.. */
/* count of points = nattack + nrelease */
};
struct iwffff_xenv {
__u8 flags;
__u8 mode;
__u8 index;
__u8 pad;
};
struct iwffff_xlayer {
__u32 stype;
__u8 flags;
__u8 velocity_mode;
__u8 layer_event;
__u8 low_range; /* range for layer based */
__u8 high_range; /* on either velocity or frequency */
__u8 pan; /* pan offset from CC1 (0 left - 127 right) */
__u8 pan_freq_scale; /* position based on frequency (0-127) */
__u8 attenuation; /* 0-127 (no corresponding midi controller) */
struct iwffff_xlfo tremolo; /* tremolo effect */
struct iwffff_xlfo vibrato; /* vibrato effect */
__u16 freq_scale; /* 0-2048, 1024 is equal to semitone scaling */
__u8 freq_center; /* center for keyboard frequency scaling */
__u8 pad;
struct iwffff_xenv penv; /* pitch envelope */
struct iwffff_xenv venv; /* volume envelope */
};
/*
* Instrument
*/
struct iwffff_xinstrument {
__u32 stype;
__u16 exclusion;
__u16 layer_type;
__u16 exclusion_group; /* 0 - none, 1-65535 */
__u8 effect1; /* effect 1 */
__u8 effect1_depth; /* 0-127 */
__u8 effect2; /* effect 2 */
__u8 effect2_depth; /* 0-127 */
};
/*
* ROM support
* InterWave ROMs are Little-Endian (x86)
*/
#define IWFFFF_ROM_HDR_SIZE 512
struct iwffff_rom_header {
__u8 iwave[8];
__u8 revision;
__u8 series_number;
__u8 series_name[16];
__u8 date[10];
__u16 vendor_revision_major;
__u16 vendor_revision_minor;
__u32 rom_size;
__u8 copyright[128];
__u8 vendor_name[64];
__u8 description[128];
};
/*
* Instrument info
*/
#define IWFFFF_INFO_LFO_VIBRATO (1<<0)
#define IWFFFF_INFO_LFO_VIBRATO_SHAPE (1<<1)
#define IWFFFF_INFO_LFO_TREMOLO (1<<2)
#define IWFFFF_INFO_LFO_TREMOLO_SHAPE (1<<3)
struct iwffff_info {
unsigned int format; /* supported format bits */
unsigned int effects; /* supported effects (1 << IWFFFF_EFFECT*) */
unsigned int lfos; /* LFO effects */
unsigned int max8_len; /* maximum 8-bit wave length */
unsigned int max16_len; /* maximum 16-bit wave length */
};
#ifdef __KERNEL__
#include "seq_instr.h"
struct snd_iwffff_ops {
void *private_data;
int (*info)(void *private_data, struct iwffff_info *info);
int (*put_sample)(void *private_data, struct iwffff_wave *wave,
char __user *data, long len, int atomic);
int (*get_sample)(void *private_data, struct iwffff_wave *wave,
char __user *data, long len, int atomic);
int (*remove_sample)(void *private_data, struct iwffff_wave *wave,
int atomic);
void (*notify)(void *private_data, struct snd_seq_kinstr *instr, int what);
struct snd_seq_kinstr_ops kops;
};
int snd_seq_iwffff_init(struct snd_iwffff_ops *ops,
void *private_data,
struct snd_seq_kinstr_ops *next);
#endif
/* typedefs for compatibility to user-space */
typedef struct iwffff_xwave iwffff_xwave_t;
typedef struct iwffff_xlfo iwffff_xlfo_t;
typedef struct iwffff_xenv_point iwffff_xenv_point_t;
typedef struct iwffff_xenv_record iwffff_xenv_record_t;
typedef struct iwffff_xenv iwffff_xenv_t;
typedef struct iwffff_xlayer iwffff_xlayer_t;
typedef struct iwffff_xinstrument iwffff_xinstrument_t;
typedef struct iwffff_rom_header iwffff_rom_header_t;
typedef struct iwffff_info iwffff_info_t;
#endif /* __SOUND_AINSTR_IW_H */

159
include/sound/ainstr_simple.h
View File

@ -1,159 +0,0 @@
/*
* Advanced Linux Sound Architecture
*
* Simple (MOD player) Instrument Format
* Copyright (c) 1994-99 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
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
*/
#ifndef __SOUND_AINSTR_SIMPLE_H
#define __SOUND_AINSTR_SIMPLE_H
#ifndef __KERNEL__
#include <asm/types.h>
#include <asm/byteorder.h>
#endif
/*
* share types (share ID 1)
*/
#define SIMPLE_SHARE_FILE 0
/*
* wave formats
*/
#define SIMPLE_WAVE_16BIT 0x0001 /* 16-bit wave */
#define SIMPLE_WAVE_UNSIGNED 0x0002 /* unsigned wave */
#define SIMPLE_WAVE_INVERT 0x0002 /* same as unsigned wave */
#define SIMPLE_WAVE_BACKWARD 0x0004 /* backward mode (maybe used for reverb or ping-ping loop) */
#define SIMPLE_WAVE_LOOP 0x0008 /* loop mode */
#define SIMPLE_WAVE_BIDIR 0x0010 /* bidirectional mode */
#define SIMPLE_WAVE_STEREO 0x0100 /* stereo wave */
#define SIMPLE_WAVE_ULAW 0x0200 /* uLaw compression mode */
/*
* instrument effects
*/
#define SIMPLE_EFFECT_NONE 0
#define SIMPLE_EFFECT_REVERB 1
#define SIMPLE_EFFECT_CHORUS 2
#define SIMPLE_EFFECT_ECHO 3
/*
* instrument info
*/
struct simple_instrument_info {
unsigned int format; /* supported format bits */
unsigned int effects; /* supported effects (1 << SIMPLE_EFFECT_*) */
unsigned int max8_len; /* maximum 8-bit wave length */
unsigned int max16_len; /* maximum 16-bit wave length */
};
/*
* Instrument
*/
struct simple_instrument {
unsigned int share_id[4]; /* share id - zero = no sharing */
unsigned int format; /* wave format */
struct {
unsigned int number; /* some other ID for this instrument */
unsigned int memory; /* begin of waveform in onboard memory */
unsigned char *ptr; /* pointer to waveform in system memory */
} address;
unsigned int size; /* size of waveform in samples */
unsigned int start; /* start offset in samples * 16 (lowest 4 bits - fraction) */
unsigned int loop_start; /* loop start offset in samples * 16 (lowest 4 bits - fraction) */
unsigned int loop_end; /* loop end offset in samples * 16 (lowest 4 bits - fraction) */
unsigned short loop_repeat; /* loop repeat - 0 = forever */
unsigned char effect1; /* effect 1 */
unsigned char effect1_depth; /* 0-127 */
unsigned char effect2; /* effect 2 */
unsigned char effect2_depth; /* 0-127 */
};
/*
*
* Kernel <-> user space
* Hardware (CPU) independent section
*
* * = zero or more
* + = one or more
*
* simple_xinstrument SIMPLE_STRU_INSTR
*
*/
#define SIMPLE_STRU_INSTR __cpu_to_be32(('I'<<24)|('N'<<16)|('S'<<8)|'T')
/*
* Instrument
*/
struct simple_xinstrument {
__u32 stype;
__u32 share_id[4]; /* share id - zero = no sharing */
__u32 format; /* wave format */
__u32 size; /* size of waveform in samples */
__u32 start; /* start offset in samples * 16 (lowest 4 bits - fraction) */
__u32 loop_start; /* bits loop start offset in samples * 16 (lowest 4 bits - fraction) */
__u32 loop_end; /* loop start offset in samples * 16 (lowest 4 bits - fraction) */
__u16 loop_repeat; /* loop repeat - 0 = forever */
__u8 effect1; /* effect 1 */
__u8 effect1_depth; /* 0-127 */
__u8 effect2; /* effect 2 */
__u8 effect2_depth; /* 0-127 */
};
#ifdef __KERNEL__
#include "seq_instr.h"
struct snd_simple_ops {
void *private_data;
int (*info)(void *private_data, struct simple_instrument_info *info);
int (*put_sample)(void *private_data, struct simple_instrument *instr,
char __user *data, long len, int atomic);
int (*get_sample)(void *private_data, struct simple_instrument *instr,
char __user *data, long len, int atomic);
int (*remove_sample)(void *private_data, struct simple_instrument *instr,
int atomic);
void (*notify)(void *private_data, struct snd_seq_kinstr *instr, int what);
struct snd_seq_kinstr_ops kops;
};
int snd_seq_simple_init(struct snd_simple_ops *ops,
void *private_data,
struct snd_seq_kinstr_ops *next);
#endif
/* typedefs for compatibility to user-space */
typedef struct simple_xinstrument simple_xinstrument_t;
#endif /* __SOUND_AINSTR_SIMPLE_H */

2
include/sound/ak4xxx-adda.h
View File

@ -68,7 +68,7 @@ struct snd_akm4xxx {
enum {
SND_AK4524, SND_AK4528, SND_AK4529,
SND_AK4355, SND_AK4358, SND_AK4381,
SND_AK5365
SND_AK5365, NON_AKM
} type;
/* (array) information of combined codecs */

242
include/sound/asequencer.h
View File

@ -110,18 +110,7 @@
#define SNDRV_SEQ_EVENT_PORT_SUBSCRIBED 66 /* ports connected */
#define SNDRV_SEQ_EVENT_PORT_UNSUBSCRIBED 67 /* ports disconnected */
/** synthesizer events
* event data type = snd_seq_eve_sample_control
*/
#define SNDRV_SEQ_EVENT_SAMPLE 70 /* sample select */
#define SNDRV_SEQ_EVENT_SAMPLE_CLUSTER 71 /* sample cluster select */
#define SNDRV_SEQ_EVENT_SAMPLE_START 72 /* voice start */
#define SNDRV_SEQ_EVENT_SAMPLE_STOP 73 /* voice stop */
#define SNDRV_SEQ_EVENT_SAMPLE_FREQ 74 /* playback frequency */
#define SNDRV_SEQ_EVENT_SAMPLE_VOLUME 75 /* volume and balance */
#define SNDRV_SEQ_EVENT_SAMPLE_LOOP 76 /* sample loop */
#define SNDRV_SEQ_EVENT_SAMPLE_POSITION 77 /* sample position */
#define SNDRV_SEQ_EVENT_SAMPLE_PRIVATE1 78 /* private (hardware dependent) event */
/* 70-89: synthesizer events - obsoleted */
/** user-defined events with fixed length
* event data type = any
@ -137,28 +126,7 @@
#define SNDRV_SEQ_EVENT_USR8 98
#define SNDRV_SEQ_EVENT_USR9 99
/** instrument layer
* variable length data can be passed directly to the driver
*/
#define SNDRV_SEQ_EVENT_INSTR_BEGIN 100 /* begin of instrument management */
#define SNDRV_SEQ_EVENT_INSTR_END 101 /* end of instrument management */
#define SNDRV_SEQ_EVENT_INSTR_INFO 102 /* instrument interface info */
#define SNDRV_SEQ_EVENT_INSTR_INFO_RESULT 103 /* result */
#define SNDRV_SEQ_EVENT_INSTR_FINFO 104 /* get format info */
#define SNDRV_SEQ_EVENT_INSTR_FINFO_RESULT 105 /* get format info */
#define SNDRV_SEQ_EVENT_INSTR_RESET 106 /* reset instrument memory */
#define SNDRV_SEQ_EVENT_INSTR_STATUS 107 /* instrument interface status */
#define SNDRV_SEQ_EVENT_INSTR_STATUS_RESULT 108 /* result */
#define SNDRV_SEQ_EVENT_INSTR_PUT 109 /* put instrument to port */
#define SNDRV_SEQ_EVENT_INSTR_GET 110 /* get instrument from port */
#define SNDRV_SEQ_EVENT_INSTR_GET_RESULT 111 /* result */
#define SNDRV_SEQ_EVENT_INSTR_FREE 112 /* free instrument(s) */
#define SNDRV_SEQ_EVENT_INSTR_LIST 113 /* instrument list */
#define SNDRV_SEQ_EVENT_INSTR_LIST_RESULT 114 /* result */
#define SNDRV_SEQ_EVENT_INSTR_CLUSTER 115 /* cluster parameters */
#define SNDRV_SEQ_EVENT_INSTR_CLUSTER_GET 116 /* get cluster parameters */
#define SNDRV_SEQ_EVENT_INSTR_CLUSTER_RESULT 117 /* result */
#define SNDRV_SEQ_EVENT_INSTR_CHANGE 118 /* instrument change */
/* 100-118: instrument layer - obsoleted */
/* 119-129: reserved */
/* 130-139: variable length events
@ -258,78 +226,6 @@ struct snd_seq_ev_ext {
void *ptr; /* pointer to data (note: maybe 64-bit) */
} __attribute__((packed));
/* Instrument cluster type */
typedef unsigned int snd_seq_instr_cluster_t;
/* Instrument type */
struct snd_seq_instr {
snd_seq_instr_cluster_t cluster;
unsigned int std; /* the upper byte means a private instrument (owner - client #) */
unsigned short bank;
unsigned short prg;
};
/* sample number */
struct snd_seq_ev_sample {
unsigned int std;
unsigned short bank;
unsigned short prg;
};
/* sample cluster */
struct snd_seq_ev_cluster {
snd_seq_instr_cluster_t cluster;
};
/* sample position */
typedef unsigned int snd_seq_position_t; /* playback position (in samples) * 16 */
/* sample stop mode */
enum {
SAMPLE_STOP_IMMEDIATELY = 0, /* terminate playing immediately */
SAMPLE_STOP_VENVELOPE = 1, /* finish volume envelope */
SAMPLE_STOP_LOOP = 2 /* terminate loop and finish wave */
};
/* sample frequency */
typedef int snd_seq_frequency_t; /* playback frequency in HZ * 16 */
/* sample volume control; if any value is set to -1 == do not change */
struct snd_seq_ev_volume {
signed short volume; /* range: 0-16383 */
signed short lr; /* left-right balance; range: 0-16383 */
signed short fr; /* front-rear balance; range: 0-16383 */
signed short du; /* down-up balance; range: 0-16383 */
};
/* simple loop redefinition */
struct snd_seq_ev_loop {
unsigned int start; /* loop start (in samples) * 16 */
unsigned int end; /* loop end (in samples) * 16 */
};
struct snd_seq_ev_sample_control {
unsigned char channel;
unsigned char unused1, unused2, unused3; /* pad */
union {
struct snd_seq_ev_sample sample;
struct snd_seq_ev_cluster cluster;
snd_seq_position_t position;
int stop_mode;
snd_seq_frequency_t frequency;
struct snd_seq_ev_volume volume;
struct snd_seq_ev_loop loop;
unsigned char raw8[8];
} param;
};
/* INSTR_BEGIN event */
struct snd_seq_ev_instr_begin {
int timeout; /* zero = forever, otherwise timeout in ms */
};
struct snd_seq_result {
int event; /* processed event type */
int result;
@ -399,8 +295,6 @@ struct snd_seq_event {
struct snd_seq_addr addr;
struct snd_seq_connect connect;
struct snd_seq_result result;
struct snd_seq_ev_instr_begin instr_begin;
struct snd_seq_ev_sample_control sample;
struct snd_seq_ev_quote quote;
} data;
};
@ -441,8 +335,6 @@ struct snd_seq_event_bounce {
#define snd_seq_ev_is_user_type(ev) ((ev)->type >= 90 && (ev)->type < 99)
/* fixed length events: 0-99 */
#define snd_seq_ev_is_fixed_type(ev) ((ev)->type < 100)
/* instrument layer events: 100-129 */
#define snd_seq_ev_is_instr_type(ev) ((ev)->type >= 100 && (ev)->type < 130)
/* variable length events: 130-139 */
#define snd_seq_ev_is_variable_type(ev) ((ev)->type >= 130 && (ev)->type < 140)
/* reserved for kernel */
@ -737,136 +629,6 @@ struct snd_seq_query_subs {
};
/*
* Instrument abstraction layer
* - based on events
*/
/* instrument types */
#define SNDRV_SEQ_INSTR_ATYPE_DATA 0 /* instrument data */
#define SNDRV_SEQ_INSTR_ATYPE_ALIAS 1 /* instrument alias */
/* instrument ASCII identifiers */
#define SNDRV_SEQ_INSTR_ID_DLS1 "DLS1"
#define SNDRV_SEQ_INSTR_ID_DLS2 "DLS2"
#define SNDRV_SEQ_INSTR_ID_SIMPLE "Simple Wave"
#define SNDRV_SEQ_INSTR_ID_SOUNDFONT "SoundFont"
#define SNDRV_SEQ_INSTR_ID_GUS_PATCH "GUS Patch"
#define SNDRV_SEQ_INSTR_ID_INTERWAVE "InterWave FFFF"
#define SNDRV_SEQ_INSTR_ID_OPL2_3 "OPL2/3 FM"
#define SNDRV_SEQ_INSTR_ID_OPL4 "OPL4"
/* instrument types */
#define SNDRV_SEQ_INSTR_TYPE0_DLS1 (1<<0) /* MIDI DLS v1 */
#define SNDRV_SEQ_INSTR_TYPE0_DLS2 (1<<1) /* MIDI DLS v2 */
#define SNDRV_SEQ_INSTR_TYPE1_SIMPLE (1<<0) /* Simple Wave */
#define SNDRV_SEQ_INSTR_TYPE1_SOUNDFONT (1<<1) /* EMU SoundFont */
#define SNDRV_SEQ_INSTR_TYPE1_GUS_PATCH (1<<2) /* Gravis UltraSound Patch */
#define SNDRV_SEQ_INSTR_TYPE1_INTERWAVE (1<<3) /* InterWave FFFF */
#define SNDRV_SEQ_INSTR_TYPE2_OPL2_3 (1<<0) /* Yamaha OPL2/3 FM */
#define SNDRV_SEQ_INSTR_TYPE2_OPL4 (1<<1) /* Yamaha OPL4 */
/* put commands */
#define SNDRV_SEQ_INSTR_PUT_CMD_CREATE 0
#define SNDRV_SEQ_INSTR_PUT_CMD_REPLACE 1
#define SNDRV_SEQ_INSTR_PUT_CMD_MODIFY 2
#define SNDRV_SEQ_INSTR_PUT_CMD_ADD 3
#define SNDRV_SEQ_INSTR_PUT_CMD_REMOVE 4
/* get commands */
#define SNDRV_SEQ_INSTR_GET_CMD_FULL 0
#define SNDRV_SEQ_INSTR_GET_CMD_PARTIAL 1
/* query flags */
#define SNDRV_SEQ_INSTR_QUERY_FOLLOW_ALIAS (1<<0)
/* free commands */
#define SNDRV_SEQ_INSTR_FREE_CMD_ALL 0
#define SNDRV_SEQ_INSTR_FREE_CMD_PRIVATE 1
#define SNDRV_SEQ_INSTR_FREE_CMD_CLUSTER 2
#define SNDRV_SEQ_INSTR_FREE_CMD_SINGLE 3
/* size of ROM/RAM */
typedef unsigned int snd_seq_instr_size_t;
/* INSTR_INFO */
struct snd_seq_instr_info {
int result; /* operation result */
unsigned int formats[8]; /* bitmap of supported formats */
int ram_count; /* count of RAM banks */
snd_seq_instr_size_t ram_sizes[16]; /* size of RAM banks */
int rom_count; /* count of ROM banks */
snd_seq_instr_size_t rom_sizes[8]; /* size of ROM banks */
char reserved[128];
};
/* INSTR_STATUS */
struct snd_seq_instr_status {
int result; /* operation result */
snd_seq_instr_size_t free_ram[16]; /* free RAM in banks */
int instrument_count; /* count of downloaded instruments */
char reserved[128];
};
/* INSTR_FORMAT_INFO */
struct snd_seq_instr_format_info {
char format[16]; /* format identifier - SNDRV_SEQ_INSTR_ID_* */
unsigned int len; /* max data length (without this structure) */
};
struct snd_seq_instr_format_info_result {
int result; /* operation result */
char format[16]; /* format identifier */
unsigned int len; /* filled data length (without this structure) */
};
/* instrument data */
struct snd_seq_instr_data {
char name[32]; /* instrument name */
char reserved[16]; /* for the future use */
int type; /* instrument type */
union {
char format[16]; /* format identifier */
struct snd_seq_instr alias;
} data;
};
/* INSTR_PUT/GET, data are stored in one block (extended), header + data */
struct snd_seq_instr_header {
union {
struct snd_seq_instr instr;
snd_seq_instr_cluster_t cluster;
} id; /* instrument identifier */
unsigned int cmd; /* get/put/free command */
unsigned int flags; /* query flags (only for get) */
unsigned int len; /* real instrument data length (without header) */
int result; /* operation result */
char reserved[16]; /* for the future */
struct snd_seq_instr_data data; /* instrument data (for put/get result) */
};
/* INSTR_CLUSTER_SET */
struct snd_seq_instr_cluster_set {
snd_seq_instr_cluster_t cluster; /* cluster identifier */
char name[32]; /* cluster name */
int priority; /* cluster priority */
char reserved[64]; /* for the future use */
};
/* INSTR_CLUSTER_GET */
struct snd_seq_instr_cluster_get {
snd_seq_instr_cluster_t cluster; /* cluster identifier */
char name[32]; /* cluster name */
int priority; /* cluster priority */
char reserved[64]; /* for the future use */
};
/*
* IOCTL commands
*/

34
include/sound/asound.h
View File

@ -95,7 +95,7 @@ enum {
SNDRV_HWDEP_IFACE_HDA, /* HD-audio */
/* Don't forget to change the following: */
SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_SB_RC
SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_HDA
};
struct snd_hwdep_info {
@ -138,7 +138,7 @@ enum {
* *
*****************************************************************************/
#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 8)
#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 9)
typedef unsigned long snd_pcm_uframes_t;
typedef signed long snd_pcm_sframes_t;
@ -354,8 +354,8 @@ struct snd_pcm_hw_params {
enum {
SNDRV_PCM_TSTAMP_NONE = 0,
SNDRV_PCM_TSTAMP_MMAP,
SNDRV_PCM_TSTAMP_LAST = SNDRV_PCM_TSTAMP_MMAP,
SNDRV_PCM_TSTAMP_ENABLE,
SNDRV_PCM_TSTAMP_LAST = SNDRV_PCM_TSTAMP_ENABLE,
};
struct snd_pcm_sw_params {
@ -363,7 +363,7 @@ struct snd_pcm_sw_params {
unsigned int period_step;
unsigned int sleep_min; /* min ticks to sleep */
snd_pcm_uframes_t avail_min; /* min avail frames for wakeup */
snd_pcm_uframes_t xfer_align; /* xfer size need to be a multiple */
snd_pcm_uframes_t xfer_align; /* obsolete: xfer size need to be a multiple */
snd_pcm_uframes_t start_threshold; /* min hw_avail frames for automatic start */
snd_pcm_uframes_t stop_threshold; /* min avail frames for automatic stop */
snd_pcm_uframes_t silence_threshold; /* min distance from noise for silence filling */
@ -434,10 +434,17 @@ struct snd_xfern {
snd_pcm_uframes_t frames;
};
enum {
SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY = 0, /* gettimeofday equivalent */
SNDRV_PCM_TSTAMP_TYPE_MONOTONIC, /* posix_clock_monotonic equivalent */
SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC,
};
enum {
SNDRV_PCM_IOCTL_PVERSION = _IOR('A', 0x00, int),
SNDRV_PCM_IOCTL_INFO = _IOR('A', 0x01, struct snd_pcm_info),
SNDRV_PCM_IOCTL_TSTAMP = _IOW('A', 0x02, int),
SNDRV_PCM_IOCTL_TTSTAMP = _IOW('A', 0x03, int),
SNDRV_PCM_IOCTL_HW_REFINE = _IOWR('A', 0x10, struct snd_pcm_hw_params),
SNDRV_PCM_IOCTL_HW_PARAMS = _IOWR('A', 0x11, struct snd_pcm_hw_params),
SNDRV_PCM_IOCTL_HW_FREE = _IO('A', 0x12),
@ -689,7 +696,7 @@ struct snd_timer_tread {
* *
****************************************************************************/
#define SNDRV_CTL_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 4)
#define SNDRV_CTL_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 5)
struct snd_ctl_card_info {
int card; /* card number */
@ -738,8 +745,7 @@ typedef int __bitwise snd_ctl_elem_iface_t;
#define SNDRV_CTL_ELEM_ACCESS_OWNER (1<<10) /* write lock owner */
#define SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK (1<<28) /* kernel use a TLV callback */
#define SNDRV_CTL_ELEM_ACCESS_USER (1<<29) /* user space element */
#define SNDRV_CTL_ELEM_ACCESS_DINDIRECT (1<<30) /* indirect access for matrix dimensions in the info structure */
#define SNDRV_CTL_ELEM_ACCESS_INDIRECT (1<<31) /* indirect access for element value in the value structure */
/* bits 30 and 31 are obsoleted (for indirect access) */
/* for further details see the ACPI and PCI power management specification */
#define SNDRV_CTL_POWER_D0 0x0000 /* full On */
@ -793,30 +799,30 @@ struct snd_ctl_elem_info {
} value;
union {
unsigned short d[4]; /* dimensions */
unsigned short *d_ptr; /* indirect */
unsigned short *d_ptr; /* indirect - obsoleted */
} dimen;
unsigned char reserved[64-4*sizeof(unsigned short)];
};
struct snd_ctl_elem_value {
struct snd_ctl_elem_id id; /* W: element ID */
unsigned int indirect: 1; /* W: use indirect pointer (xxx_ptr member) */
unsigned int indirect: 1; /* W: indirect access - obsoleted */
union {
union {
long value[128];
long *value_ptr;
long *value_ptr; /* obsoleted */
} integer;
union {
long long value[64];
long long *value_ptr;
long long *value_ptr; /* obsoleted */
} integer64;
union {
unsigned int item[128];
unsigned int *item_ptr;
unsigned int *item_ptr; /* obsoleted */
} enumerated;
union {
unsigned char data[512];
unsigned char *data_ptr;
unsigned char *data_ptr; /* obsoleted */
} bytes;
struct snd_aes_iec958 iec958;
} value; /* RO */

19
include/sound/asound_fm.h
View File

@ -104,6 +104,8 @@ struct snd_dm_fm_params {
#define SNDRV_DM_FM_IOCTL_SET_MODE _IOW('H', 0x25, int)
/* for OPL3 only */
#define SNDRV_DM_FM_IOCTL_SET_CONNECTION _IOW('H', 0x26, int)
/* SBI patch management */
#define SNDRV_DM_FM_IOCTL_CLEAR_PATCHES _IO ('H', 0x40)
#define SNDRV_DM_FM_OSS_IOCTL_RESET 0x20
#define SNDRV_DM_FM_OSS_IOCTL_PLAY_NOTE 0x21
@ -112,4 +114,21 @@ struct snd_dm_fm_params {
#define SNDRV_DM_FM_OSS_IOCTL_SET_MODE 0x24
#define SNDRV_DM_FM_OSS_IOCTL_SET_OPL 0x25
/*
* Patch Record - fixed size for write
*/
#define FM_KEY_SBI "SBI\032"
#define FM_KEY_2OP "2OP\032"
#define FM_KEY_4OP "4OP\032"
struct sbi_patch {
unsigned char prog;
unsigned char bank;
char key[4];
char name[25];
char extension[7];
unsigned char data[32];
};
#endif /* __SOUND_ASOUND_FM_H */

10
include/sound/core.h
View File

@ -22,12 +22,22 @@
*
*/
#include <linux/module.h>
#include <linux/sched.h> /* wake_up() */
#include <linux/mutex.h> /* struct mutex */
#include <linux/rwsem.h> /* struct rw_semaphore */
#include <linux/pm.h> /* pm_message_t */
#include <linux/device.h>
/* number of supported soundcards */
#ifdef CONFIG_SND_DYNAMIC_MINORS
#define SNDRV_CARDS 32
#else
#define SNDRV_CARDS 8 /* don't change - minor numbers */
#endif
#define CONFIG_SND_MAJOR 116 /* standard configuration */
/* forward declarations */
#ifdef CONFIG_PCI
struct pci_dev;

2
include/sound/cs4231-regs.h
View File

@ -45,7 +45,7 @@
#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_MISC_INFO 0x0c /* miscellaneous 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 */

3
include/sound/cs46xx.h
View File

@ -1708,9 +1708,6 @@ struct snd_cs46xx {
struct gameport *gameport;
#ifdef CONFIG_SND_CS46XX_DEBUG_GPIO
int current_gpio;
#endif
#ifdef CONFIG_SND_CS46XX_NEW_DSP
struct mutex spos_mutex;

52
include/sound/driver.h
View File

@ -1,51 +1 @@
#ifndef __SOUND_DRIVER_H
#define __SOUND_DRIVER_H
/*
* Main header file for the ALSA driver
* Copyright (c) 1994-2000 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
* 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
*
*/
#ifdef ALSA_BUILD
#include "config.h"
#endif
/* number of supported soundcards */
#ifdef CONFIG_SND_DYNAMIC_MINORS
#define SNDRV_CARDS 32
#else
#define SNDRV_CARDS 8 /* don't change - minor numbers */
#endif
#ifndef CONFIG_SND_MAJOR /* standard configuration */
#define CONFIG_SND_MAJOR 116
#endif
#ifndef CONFIG_SND_DEBUG
#undef CONFIG_SND_DEBUG_MEMORY
#endif
#ifdef ALSA_BUILD
#include "adriver.h"
#endif
#include <linux/module.h>
#endif /* __SOUND_DRIVER_H */
#warning "This file is deprecated"

231
include/sound/emu10k1.h
View File

@ -1120,6 +1120,99 @@
/************************************************************************************************/
/* EMU1010m HANA Destinations */
/************************************************************************************************/
/* Hana, original 1010,1212,1820 using Alice2
* Destiniations for SRATEX = 1X rates: 44.1 kHz or 48 kHz
* 0x00, 0x00-0x0f: 16 EMU32 channels to Alice2
* 0x01, 0x10-0x1f: 32 Elink channels to Audio Dock
* 0x01, 0x00: Dock DAC 1 Left
* 0x01, 0x04: Dock DAC 1 Right
* 0x01, 0x08: Dock DAC 2 Left
* 0x01, 0x0c: Dock DAC 2 Right
* 0x01, 0x10: Dock DAC 3 Left
* 0x01, 0x12: PHONES Left
* 0x01, 0x14: Dock DAC 3 Right
* 0x01, 0x16: PHONES Right
* 0x01, 0x18: Dock DAC 4 Left
* 0x01, 0x1a: S/PDIF Left
* 0x01, 0x1c: Dock DAC 4 Right
* 0x01, 0x1e: S/PDIF Right
* 0x02, 0x00: Hana S/PDIF Left
* 0x02, 0x01: Hana S/PDIF Right
* 0x03, 0x00: Hanoa DAC Left
* 0x03, 0x01: Hanoa DAC Right
* 0x04, 0x00-0x07: Hana ADAT
* 0x05, 0x00: I2S0 Left to Alice2
* 0x05, 0x01: I2S0 Right to Alice2
* 0x06, 0x00: I2S0 Left to Alice2
* 0x06, 0x01: I2S0 Right to Alice2
* 0x07, 0x00: I2S0 Left to Alice2
* 0x07, 0x01: I2S0 Right to Alice2
*
* Hana2 never released, but used Tina
* Not needed.
*
* Hana3, rev2 1010,1212,1616 using Tina
* Destinations for SRATEX = 1X rates: 44.1 kHz or 48 kHz
* 0x00, 0x00-0x0f: 16 EMU32A channels to Tina
* 0x01, 0x10-0x1f: 32 EDI channels to Micro Dock
* 0x01, 0x00: Dock DAC 1 Left
* 0x01, 0x04: Dock DAC 1 Right
* 0x01, 0x08: Dock DAC 2 Left
* 0x01, 0x0c: Dock DAC 2 Right
* 0x01, 0x10: Dock DAC 3 Left
* 0x01, 0x12: Dock S/PDIF Left
* 0x01, 0x14: Dock DAC 3 Right
* 0x01, 0x16: Dock S/PDIF Right
* 0x01, 0x18-0x1f: Dock ADAT 0-7
* 0x02, 0x00: Hana3 S/PDIF Left
* 0x02, 0x01: Hana3 S/PDIF Right
* 0x03, 0x00: Hanoa DAC Left
* 0x03, 0x01: Hanoa DAC Right
* 0x04, 0x00-0x07: Hana3 ADAT 0-7
* 0x05, 0x00-0x0f: 16 EMU32B channels to Tina
* 0x06-0x07: Not used
*
* HanaLite, rev1 0404 using Alice2
* Destiniations for SRATEX = 1X rates: 44.1 kHz or 48 kHz
* 0x00, 0x00-0x0f: 16 EMU32 channels to Alice2
* 0x01: Not used
* 0x02, 0x00: S/PDIF Left
* 0x02, 0x01: S/PDIF Right
* 0x03, 0x00: DAC Left
* 0x03, 0x01: DAC Right
* 0x04-0x07: Not used
*
* HanaLiteLite, rev2 0404 using Alice2
* Destiniations for SRATEX = 1X rates: 44.1 kHz or 48 kHz
* 0x00, 0x00-0x0f: 16 EMU32 channels to Alice2
* 0x01: Not used
* 0x02, 0x00: S/PDIF Left
* 0x02, 0x01: S/PDIF Right
* 0x03, 0x00: DAC Left
* 0x03, 0x01: DAC Right
* 0x04-0x07: Not used
*
* Mana, Cardbus 1616 using Tina2
* Destinations for SRATEX = 1X rates: 44.1 kHz or 48 kHz
* 0x00, 0x00-0x0f: 16 EMU32A channels to Tina2
* 0x01, 0x10-0x1f: 32 EDI channels to Micro Dock
* 0x01, 0x00: Dock DAC 1 Left
* 0x01, 0x04: Dock DAC 1 Right
* 0x01, 0x08: Dock DAC 2 Left
* 0x01, 0x0c: Dock DAC 2 Right
* 0x01, 0x10: Dock DAC 3 Left
* 0x01, 0x12: Dock S/PDIF Left
* 0x01, 0x14: Dock DAC 3 Right
* 0x01, 0x16: Dock S/PDIF Right
* 0x01, 0x18-0x1f: Dock ADAT 0-7
* 0x02: Not used
* 0x03, 0x00: Mana DAC Left
* 0x03, 0x01: Mana DAC Right
* 0x04, 0x00-0x0f: 16 EMU32B channels to Tina2
* 0x05-0x07: Not used
*
*
*/
/* 32-bit destinations of signal in the Hana FPGA. Destinations are either
* physical outputs of Hana, or outputs going to Alice2 (audigy) for capture
* - 16 x EMU_DST_ALICE2_EMU32_X.
@ -1206,9 +1299,122 @@
#define EMU_DST_ALICE_I2S2_LEFT 0x0700 /* Alice2 I2S2 Left */
#define EMU_DST_ALICE_I2S2_RIGHT 0x0701 /* Alice2 I2S2 Right */
/* Additional destinations for 1616(M)/Microdock */
/* Microdock S/PDIF OUT Left, 1st or 48kHz only */
#define EMU_DST_MDOCK_SPDIF_LEFT1 0x0112
/* Microdock S/PDIF OUT Left, 2nd or 96kHz */
#define EMU_DST_MDOCK_SPDIF_LEFT2 0x0113
/* Microdock S/PDIF OUT Right, 1st or 48kHz only */
#define EMU_DST_MDOCK_SPDIF_RIGHT1 0x0116
/* Microdock S/PDIF OUT Right, 2nd or 96kHz */
#define EMU_DST_MDOCK_SPDIF_RIGHT2 0x0117
/* Microdock S/PDIF ADAT 8 channel out +8 to +f */
#define EMU_DST_MDOCK_ADAT 0x0118
/* Headphone jack on 1010 cardbus? 44.1/48kHz only? */
#define EMU_DST_MANA_DAC_LEFT 0x0300
/* Headphone jack on 1010 cardbus? 44.1/48kHz only? */
#define EMU_DST_MANA_DAC_RIGHT 0x0301
/************************************************************************************************/
/* EMU1010m HANA Sources */
/************************************************************************************************/
/* Hana, original 1010,1212,1820 using Alice2
* Sources SRATEX = 1X rates: 44.1 kHz or 48 kHz
* 0x00,0x00-0x1f: Silence
* 0x01, 0x10-0x1f: 32 Elink channels from Audio Dock
* 0x01, 0x00: Dock Mic A
* 0x01, 0x04: Dock Mic B
* 0x01, 0x08: Dock ADC 1 Left
* 0x01, 0x0c: Dock ADC 1 Right
* 0x01, 0x10: Dock ADC 2 Left
* 0x01, 0x14: Dock ADC 2 Right
* 0x01, 0x18: Dock ADC 3 Left
* 0x01, 0x1c: Dock ADC 3 Right
* 0x02, 0x00: Hana ADC Left
* 0x02, 0x01: Hana ADC Right
* 0x03, 0x00-0x0f: 16 inputs from Alice2 Emu32A output
* 0x03, 0x10-0x1f: 16 inputs from Alice2 Emu32B output
* 0x04, 0x00-0x07: Hana ADAT
* 0x05, 0x00: Hana S/PDIF Left
* 0x05, 0x01: Hana S/PDIF Right
* 0x06-0x07: Not used
*
* Hana2 never released, but used Tina
* Not needed.
*
* Hana3, rev2 1010,1212,1616 using Tina
* Sources SRATEX = 1X rates: 44.1 kHz or 48 kHz
* 0x00,0x00-0x1f: Silence
* 0x01, 0x10-0x1f: 32 Elink channels from Audio Dock
* 0x01, 0x00: Dock Mic A
* 0x01, 0x04: Dock Mic B
* 0x01, 0x08: Dock ADC 1 Left
* 0x01, 0x0c: Dock ADC 1 Right
* 0x01, 0x10: Dock ADC 2 Left
* 0x01, 0x12: Dock S/PDIF Left
* 0x01, 0x14: Dock ADC 2 Right
* 0x01, 0x16: Dock S/PDIF Right
* 0x01, 0x18-0x1f: Dock ADAT 0-7
* 0x01, 0x18: Dock ADC 3 Left
* 0x01, 0x1c: Dock ADC 3 Right
* 0x02, 0x00: Hanoa ADC Left
* 0x02, 0x01: Hanoa ADC Right
* 0x03, 0x00-0x0f: 16 inputs from Tina Emu32A output
* 0x03, 0x10-0x1f: 16 inputs from Tina Emu32B output
* 0x04, 0x00-0x07: Hana3 ADAT
* 0x05, 0x00: Hana3 S/PDIF Left
* 0x05, 0x01: Hana3 S/PDIF Right
* 0x06-0x07: Not used
*
* HanaLite, rev1 0404 using Alice2
* Sources SRATEX = 1X rates: 44.1 kHz or 48 kHz
* 0x00,0x00-0x1f: Silence
* 0x01: Not used
* 0x02, 0x00: ADC Left
* 0x02, 0x01: ADC Right
* 0x03, 0x00-0x0f: 16 inputs from Alice2 Emu32A output
* 0x03, 0x10-0x1f: 16 inputs from Alice2 Emu32B output
* 0x04: Not used
* 0x05, 0x00: S/PDIF Left
* 0x05, 0x01: S/PDIF Right
* 0x06-0x07: Not used
*
* HanaLiteLite, rev2 0404 using Alice2
* Sources SRATEX = 1X rates: 44.1 kHz or 48 kHz
* 0x00,0x00-0x1f: Silence
* 0x01: Not used
* 0x02, 0x00: ADC Left
* 0x02, 0x01: ADC Right
* 0x03, 0x00-0x0f: 16 inputs from Alice2 Emu32A output
* 0x03, 0x10-0x1f: 16 inputs from Alice2 Emu32B output
* 0x04: Not used
* 0x05, 0x00: S/PDIF Left
* 0x05, 0x01: S/PDIF Right
* 0x06-0x07: Not used
*
* Mana, Cardbus 1616 using Tina2
* Sources SRATEX = 1X rates: 44.1 kHz or 48 kHz
* 0x00,0x00-0x1f: Silence
* 0x01, 0x10-0x1f: 32 Elink channels from Audio Dock
* 0x01, 0x00: Dock Mic A
* 0x01, 0x04: Dock Mic B
* 0x01, 0x08: Dock ADC 1 Left
* 0x01, 0x0c: Dock ADC 1 Right
* 0x01, 0x10: Dock ADC 2 Left
* 0x01, 0x12: Dock S/PDIF Left
* 0x01, 0x14: Dock ADC 2 Right
* 0x01, 0x16: Dock S/PDIF Right
* 0x01, 0x18-0x1f: Dock ADAT 0-7
* 0x01, 0x18: Dock ADC 3 Left
* 0x01, 0x1c: Dock ADC 3 Right
* 0x02: Not used
* 0x03, 0x00-0x0f: 16 inputs from Tina Emu32A output
* 0x03, 0x10-0x1f: 16 inputs from Tina Emu32B output
* 0x04-0x07: Not used
*
*/
/* 32-bit sources of signal in the Hana FPGA. The sources are routed to
* destinations using mixer control for each destination - see emumixer.c
* Sources are either physical inputs of FPGA,
@ -1263,6 +1469,19 @@
#define EMU_SRC_HANA_SPDIF_LEFT2 0x0502 /* Hana SPDIF Left, 2nd or 96kHz */
#define EMU_SRC_HANA_SPDIF_RIGHT1 0x0501 /* Hana SPDIF Right, 1st or 48kHz only */
#define EMU_SRC_HANA_SPDIF_RIGHT2 0x0503 /* Hana SPDIF Right, 2nd or 96kHz */
/* Additional inputs for 1616(M)/Microdock */
/* Microdock S/PDIF Left, 1st or 48kHz only */
#define EMU_SRC_MDOCK_SPDIF_LEFT1 0x0112
/* Microdock S/PDIF Left, 2nd or 96kHz */
#define EMU_SRC_MDOCK_SPDIF_LEFT2 0x0113
/* Microdock S/PDIF Right, 1st or 48kHz only */
#define EMU_SRC_MDOCK_SPDIF_RIGHT1 0x0116
/* Microdock S/PDIF Right, 2nd or 96kHz */
#define EMU_SRC_MDOCK_SPDIF_RIGHT2 0x0117
/* Microdock ADAT 8 channel in +8 to +f */
#define EMU_SRC_MDOCK_ADAT 0x0118
/* 0x600 and 0x700 no used */
/* ------------------- STRUCTURES -------------------- */
@ -1423,6 +1642,14 @@ struct snd_emu10k1_midi {
void (*interrupt)(struct snd_emu10k1 *emu, unsigned int status);
};
enum {
EMU_MODEL_SB,
EMU_MODEL_EMU1010,
EMU_MODEL_EMU1010B,
EMU_MODEL_EMU1616,
EMU_MODEL_EMU0404,
};
struct snd_emu_chip_details {
u32 vendor;
u32 device;
@ -1439,7 +1666,7 @@ struct snd_emu_chip_details {
unsigned char spdif_bug; /* Has Spdif phasing bug */
unsigned char ac97_chip; /* Has an AC97 chip: 1 = mandatory, 2 = optional */
unsigned char ecard; /* APS EEPROM */
unsigned char emu1010; /* EMU 1010m card */
unsigned char emu_model; /* EMU model type */
unsigned char spi_dac; /* SPI interface for DAC */
unsigned char i2c_adc; /* I2C interface for ADC */
unsigned char adc_1361t; /* Use Philips 1361T ADC */
@ -1515,6 +1742,8 @@ struct snd_emu10k1 {
spinlock_t reg_lock;
spinlock_t emu_lock;
spinlock_t voice_lock;
spinlock_t spi_lock; /* serialises access to spi port */
spinlock_t i2c_lock; /* serialises access to i2c port */
struct snd_emu10k1_voice voices[NUM_G];
struct snd_emu10k1_voice p16v_voices[4];

63
include/sound/gus.h
View File

@ -27,13 +27,8 @@
#include "timer.h"
#include "seq_midi_emul.h"
#include "seq_device.h"
#include "ainstr_iw.h"
#include "ainstr_gf1.h"
#include "ainstr_simple.h"
#include <asm/io.h>
#define SNDRV_SEQ_DEV_ID_GUS "gus-synth"
/* IO ports */
#define GUSP(gus, x) ((gus)->gf1.port + SNDRV_g_u_s_##x)
@ -234,16 +229,6 @@ struct snd_gus_port {
struct snd_gus_voice;
struct snd_gus_sample_ops {
void (*sample_start)(struct snd_gus_card *gus, struct snd_gus_voice *voice, snd_seq_position_t position);
void (*sample_stop)(struct snd_gus_card *gus, struct snd_gus_voice *voice, int mode);
void (*sample_freq)(struct snd_gus_card *gus, struct snd_gus_voice *voice, snd_seq_frequency_t freq);
void (*sample_volume)(struct snd_gus_card *gus, struct snd_gus_voice *voice, struct snd_seq_ev_volume *volume);
void (*sample_loop)(struct snd_gus_card *card, struct snd_gus_voice *voice, struct snd_seq_ev_loop *loop);
void (*sample_pos)(struct snd_gus_card *card, struct snd_gus_voice *voice, snd_seq_position_t position);
void (*sample_private1)(struct snd_gus_card *card, struct snd_gus_voice *voice, unsigned char *data);
};
#define SNDRV_GF1_VOICE_TYPE_PCM 0
#define SNDRV_GF1_VOICE_TYPE_SYNTH 1
#define SNDRV_GF1_VOICE_TYPE_MIDI 2
@ -284,12 +269,8 @@ struct snd_gus_voice {
struct snd_gus_sample_ops *sample_ops;
struct snd_seq_instr instr;
/* running status / registers */
struct snd_seq_ev_volume sample_volume;
unsigned short fc_register;
unsigned short fc_lfo;
unsigned short gf1_volume;
@ -382,10 +363,6 @@ struct snd_gf1 {
int seq_client;
struct snd_gus_port seq_ports[4];
struct snd_seq_kinstr_list *ilist;
struct snd_iwffff_ops iwffff_ops;
struct snd_gf1_ops gf1_ops;
struct snd_simple_ops simple_ops;
/* timer */
@ -458,8 +435,6 @@ struct snd_gus_card {
struct snd_rawmidi_substream *midi_substream_output;
struct snd_rawmidi_substream *midi_substream_input;
struct snd_seq_device *seq_dev;
spinlock_t reg_lock;
spinlock_t voice_alloc;
spinlock_t active_voice_lock;
@ -647,48 +622,10 @@ void snd_gus_irq_profile_init(struct snd_gus_card *gus);
int snd_gf1_rawmidi_new(struct snd_gus_card * gus, int device, struct snd_rawmidi **rrawmidi);
#if 0
extern void snd_engine_instrument_register(unsigned short mode,
struct _SND_INSTRUMENT_VOICE_COMMANDS *voice_cmds,
struct _SND_INSTRUMENT_NOTE_COMMANDS *note_cmds,
struct _SND_INSTRUMENT_CHANNEL_COMMANDS *channel_cmds);
extern int snd_engine_instrument_register_ask(unsigned short mode);
#endif
/* gus_dram.c */
int snd_gus_dram_write(struct snd_gus_card *gus, char __user *ptr,
unsigned int addr, unsigned int size);
int snd_gus_dram_read(struct snd_gus_card *gus, char __user *ptr,
unsigned int addr, unsigned int size, int rom);
#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE)
/* gus_sample.c */
void snd_gus_sample_event(struct snd_seq_event *ev, struct snd_gus_port *p);
/* gus_simple.c */
void snd_gf1_simple_init(struct snd_gus_voice *voice);
/* gus_instr.c */
int snd_gus_iwffff_put_sample(void *private_data, struct iwffff_wave *wave,
char __user *data, long len, int atomic);
int snd_gus_iwffff_get_sample(void *private_data, struct iwffff_wave *wave,
char __user *data, long len, int atomic);
int snd_gus_iwffff_remove_sample(void *private_data, struct iwffff_wave *wave,
int atomic);
int snd_gus_gf1_put_sample(void *private_data, struct gf1_wave *wave,
char __user *data, long len, int atomic);
int snd_gus_gf1_get_sample(void *private_data, struct gf1_wave *wave,
char __user *data, long len, int atomic);
int snd_gus_gf1_remove_sample(void *private_data, struct gf1_wave *wave,
int atomic);
int snd_gus_simple_put_sample(void *private_data, struct simple_instrument *instr,
char __user *data, long len, int atomic);
int snd_gus_simple_get_sample(void *private_data, struct simple_instrument *instr,
char __user *data, long len, int atomic);
int snd_gus_simple_remove_sample(void *private_data, struct simple_instrument *instr,
int atomic);
#endif /* CONFIG_SND_SEQUENCER */
#endif /* __SOUND_GUS_H */

2
include/sound/info.h
View File

@ -100,8 +100,10 @@ int snd_info_minor_unregister(void);
extern struct snd_info_entry *snd_seq_root;
#ifdef CONFIG_SND_OSSEMUL
extern struct snd_info_entry *snd_oss_root;
void snd_card_info_read_oss(struct snd_info_buffer *buffer);
#else
#define snd_oss_root NULL
static inline void snd_card_info_read_oss(struct snd_info_buffer *buffer) {}
#endif
int snd_iprintf(struct snd_info_buffer * buffer, char *fmt,...) __attribute__ ((format (printf, 2, 3)));

78
include/sound/opl3.h
View File

@ -51,19 +51,16 @@
*
*/
#include "driver.h"
#include <linux/time.h>
#include <linux/mutex.h>
#include "core.h"
#include "hwdep.h"
#include "timer.h"
#include "seq_midi_emul.h"
#include <sound/core.h>
#include <sound/hwdep.h>
#include <sound/timer.h>
#include <sound/seq_midi_emul.h>
#ifdef CONFIG_SND_SEQUENCER_OSS
#include "seq_oss.h"
#include "seq_oss_legacy.h"
#include <sound/seq_oss.h>
#include <sound/seq_oss_legacy.h>
#endif
#include "seq_device.h"
#include "ainstr_fm.h"
#include <sound/seq_device.h>
#include <sound/asound_fm.h>
/*
* Register numbers for the global registers
@ -239,6 +236,47 @@
struct snd_opl3;
/*
* Instrument record, aka "Patch"
*/
/* FM operator */
struct fm_operator {
unsigned char am_vib;
unsigned char ksl_level;
unsigned char attack_decay;
unsigned char sustain_release;
unsigned char wave_select;
} __attribute__((packed));
/* Instrument data */
struct fm_instrument {
struct fm_operator op[4];
unsigned char feedback_connection[2];
unsigned char echo_delay;
unsigned char echo_atten;
unsigned char chorus_spread;
unsigned char trnsps;
unsigned char fix_dur;
unsigned char modes;
unsigned char fix_key;
};
/* type */
#define FM_PATCH_OPL2 0x01 /* OPL2 2 operators FM instrument */
#define FM_PATCH_OPL3 0x02 /* OPL3 4 operators FM instrument */
/* Instrument record */
struct fm_patch {
unsigned char prog;
unsigned char bank;
unsigned char type;
struct fm_instrument inst;
char name[24];
struct fm_patch *next;
};
/*
* A structure to keep track of each hardware voice
*/
@ -277,9 +315,9 @@ struct snd_opl3 {
void *private_data;
void (*private_free)(struct snd_opl3 *);
struct snd_hwdep *hwdep;
spinlock_t reg_lock;
struct snd_card *card; /* The card that this belongs to */
int used; /* usage flag - exclusive */
unsigned char fm_mode; /* OPL mode, see SNDRV_DM_FM_MODE_XXX */
unsigned char rhythm; /* percussion mode flag */
unsigned char max_voices; /* max number of voices */
@ -297,8 +335,8 @@ struct snd_opl3 {
struct snd_midi_channel_set * oss_chset;
#endif
struct snd_seq_kinstr_ops fm_ops;
struct snd_seq_kinstr_list *ilist;
#define OPL3_PATCH_HASH_SIZE 32
struct fm_patch *patch_table[OPL3_PATCH_HASH_SIZE];
struct snd_opl3_voice voices[MAX_OPL3_VOICES]; /* Voices (OPL3 'channel') */
int use_time; /* allocation counter */
@ -312,7 +350,6 @@ struct snd_opl3 {
int sys_timer_status; /* system timer run status */
spinlock_t sys_timer_lock; /* Lock for system timer access */
#endif
struct mutex access_mutex; /* locking */
};
/* opl3.c */
@ -333,8 +370,19 @@ int snd_opl3_hwdep_new(struct snd_opl3 * opl3, int device, int seq_device,
int snd_opl3_open(struct snd_hwdep * hw, struct file *file);
int snd_opl3_ioctl(struct snd_hwdep * hw, struct file *file,
unsigned int cmd, unsigned long arg);
long snd_opl3_write(struct snd_hwdep *hw, const char __user *buf, long count,
loff_t *offset);
int snd_opl3_release(struct snd_hwdep * hw, struct file *file);
void snd_opl3_reset(struct snd_opl3 * opl3);
int snd_opl3_load_patch(struct snd_opl3 *opl3,
int prog, int bank, int type,
const char *name,
const unsigned char *ext,
const unsigned char *data);
struct fm_patch *snd_opl3_find_patch(struct snd_opl3 *opl3, int prog, int bank,
int create_patch);
void snd_opl3_clear_patches(struct snd_opl3 *opl3);
#endif /* __SOUND_OPL3_H */

18
include/sound/pcm.h
View File

@ -274,7 +274,6 @@ struct snd_pcm_runtime {
snd_pcm_uframes_t period_size; /* period size */
unsigned int periods; /* periods */
snd_pcm_uframes_t buffer_size; /* buffer size */
unsigned int tick_time; /* tick time */
snd_pcm_uframes_t min_align; /* Min alignment for the format */
size_t byte_align;
unsigned int frame_bits;
@ -286,8 +285,6 @@ struct snd_pcm_runtime {
/* -- SW params -- */
int tstamp_mode; /* mmap timestamp is updated */
unsigned int period_step;
unsigned int sleep_min; /* min ticks to sleep */
snd_pcm_uframes_t xfer_align; /* xfer size need to be a multiple */
snd_pcm_uframes_t start_threshold;
snd_pcm_uframes_t stop_threshold;
snd_pcm_uframes_t silence_threshold; /* Silence filling happens when
@ -306,7 +303,6 @@ struct snd_pcm_runtime {
/* -- locking / scheduling -- */
wait_queue_head_t sleep;
struct timer_list tick_timer;
struct fasync_struct *fasync;
/* -- private section -- */
@ -323,6 +319,7 @@ struct snd_pcm_runtime {
/* -- timer -- */
unsigned int timer_resolution; /* timer resolution */
int tstamp_type; /* timestamp type */
/* -- DMA -- */
unsigned char *dma_area; /* DMA area */
@ -810,7 +807,6 @@ static inline const struct snd_interval *hw_param_interval_c(const struct snd_pc
#define params_periods(p) hw_param_interval((p), SNDRV_PCM_HW_PARAM_PERIODS)->min
#define params_buffer_size(p) hw_param_interval((p), SNDRV_PCM_HW_PARAM_BUFFER_SIZE)->min
#define params_buffer_bytes(p) hw_param_interval((p), SNDRV_PCM_HW_PARAM_BUFFER_BYTES)->min
#define params_tick_time(p) hw_param_interval((p), SNDRV_PCM_HW_PARAM_TICK_TIME)->min
int snd_interval_refine(struct snd_interval *i, const struct snd_interval *v);
@ -908,9 +904,6 @@ int snd_pcm_capture_xrun_check(struct snd_pcm_substream *substream);
int snd_pcm_playback_xrun_asap(struct snd_pcm_substream *substream);
int snd_pcm_capture_xrun_asap(struct snd_pcm_substream *substream);
void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_uframes_t new_hw_ptr);
void snd_pcm_tick_prepare(struct snd_pcm_substream *substream);
void snd_pcm_tick_set(struct snd_pcm_substream *substream, unsigned long ticks);
void snd_pcm_tick_elapsed(struct snd_pcm_substream *substream);
void snd_pcm_period_elapsed(struct snd_pcm_substream *substream);
snd_pcm_sframes_t snd_pcm_lib_write(struct snd_pcm_substream *substream,
const void __user *buf,
@ -952,6 +945,15 @@ void snd_pcm_timer_resolution_change(struct snd_pcm_substream *substream);
void snd_pcm_timer_init(struct snd_pcm_substream *substream);
void snd_pcm_timer_done(struct snd_pcm_substream *substream);
static inline void snd_pcm_gettime(struct snd_pcm_runtime *runtime,
struct timespec *tv)
{
if (runtime->tstamp_type == SNDRV_PCM_TSTAMP_TYPE_MONOTONIC)
do_posix_clock_monotonic_gettime(tv);
else
getnstimeofday(tv);
}
/*
* Memory
*/

110
include/sound/seq_instr.h
View File

@ -1,110 +0,0 @@
#ifndef __SOUND_SEQ_INSTR_H
#define __SOUND_SEQ_INSTR_H
/*
* Main kernel header file for the ALSA sequencer
* 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
* 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 "seq_kernel.h"
/* Instrument cluster */
struct snd_seq_kcluster {
snd_seq_instr_cluster_t cluster;
char name[32];
int priority;
struct snd_seq_kcluster *next;
};
/* return pointer to private data */
#define KINSTR_DATA(kinstr) (void *)(((char *)kinstr) + sizeof(struct snd_seq_kinstr))
/* Instrument structure */
struct snd_seq_kinstr {
struct snd_seq_instr instr;
char name[32];
int type; /* instrument type */
int use; /* use count */
int busy; /* not useable */
int add_len; /* additional length */
struct snd_seq_kinstr_ops *ops; /* operations */
struct snd_seq_kinstr *next;
};
#define SNDRV_SEQ_INSTR_HASH_SIZE 32
/* Instrument flags */
#define SNDRV_SEQ_INSTR_FLG_DIRECT (1<<0) /* accept only direct events */
/* List of all instruments */
struct snd_seq_kinstr_list {
struct snd_seq_kinstr *hash[SNDRV_SEQ_INSTR_HASH_SIZE];
int count; /* count of all instruments */
struct snd_seq_kcluster *chash[SNDRV_SEQ_INSTR_HASH_SIZE];
int ccount; /* count of all clusters */
int owner; /* current owner of the instrument list */
unsigned int flags;
spinlock_t lock;
spinlock_t ops_lock;
struct mutex ops_mutex;
unsigned long ops_flags;
};
#define SNDRV_SEQ_INSTR_NOTIFY_REMOVE 0
#define SNDRV_SEQ_INSTR_NOTIFY_CHANGE 1
struct snd_seq_kinstr_ops {
void *private_data;
long add_len; /* additional length */
char *instr_type;
int (*info)(void *private_data, char *info_data, long len);
int (*put)(void *private_data, struct snd_seq_kinstr *kinstr,
char __user *instr_data, long len, int atomic, int cmd);
int (*get)(void *private_data, struct snd_seq_kinstr *kinstr,
char __user *instr_data, long len, int atomic, int cmd);
int (*get_size)(void *private_data, struct snd_seq_kinstr *kinstr, long *size);
int (*remove)(void *private_data, struct snd_seq_kinstr *kinstr, int atomic);
void (*notify)(void *private_data, struct snd_seq_kinstr *kinstr, int what);
struct snd_seq_kinstr_ops *next;
};
/* instrument operations */
struct snd_seq_kinstr_list *snd_seq_instr_list_new(void);
void snd_seq_instr_list_free(struct snd_seq_kinstr_list **list);
int snd_seq_instr_list_free_cond(struct snd_seq_kinstr_list *list,
struct snd_seq_instr_header *ifree,
int client,
int atomic);
struct snd_seq_kinstr *snd_seq_instr_find(struct snd_seq_kinstr_list *list,
struct snd_seq_instr *instr,
int exact,
int follow_alias);
void snd_seq_instr_free_use(struct snd_seq_kinstr_list *list,
struct snd_seq_kinstr *instr);
int snd_seq_instr_event(struct snd_seq_kinstr_ops *ops,
struct snd_seq_kinstr_list *list,
struct snd_seq_event *ev,
int client,
int atomic,
int hop);
#endif /* __SOUND_SEQ_INSTR_H */

29
include/sound/soc-dapm.h
View File

@ -22,7 +22,7 @@
#define SND_SOC_NOPM -1
/*
* SoC dynamic audio power managment
* SoC dynamic audio power management
*
* We can have upto 4 power domains
* 1. Codec domain - VREF, VMID
@ -131,18 +131,34 @@
.shift = wshift, .invert = winvert}
/* dapm kcontrol types */
#define SOC_DAPM_SINGLE(xname, reg, shift, mask, invert) \
#define SOC_DAPM_SINGLE(xname, reg, shift, max, invert) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = snd_soc_info_volsw, \
.get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \
.private_value = SOC_SINGLE_VALUE(reg, shift, mask, invert) }
#define SOC_DAPM_DOUBLE(xname, reg, shift_left, shift_right, mask, invert, \
.private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) }
#define SOC_DAPM_DOUBLE(xname, reg, shift_left, shift_right, max, invert, \
power) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
.info = snd_soc_info_volsw, \
.get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \
.private_value = (reg) | ((shift_left) << 8) | ((shift_right) << 12) |\
((mask) << 16) | ((invert) << 24) }
((max) << 16) | ((invert) << 24) }
#define SOC_DAPM_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = snd_soc_info_volsw, \
.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE,\
.tlv.p = (tlv_array), \
.get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \
.private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) }
#define SOC_DAPM_DOUBLE_TLV(xname, reg, shift_left, shift_right, max, invert, \
power, tlv_array) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE,\
.tlv.p = (tlv_array), \
.info = snd_soc_info_volsw, \
.get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \
.private_value = (reg) | ((shift_left) << 8) | ((shift_right) << 12) |\
((max) << 16) | ((invert) << 24) }
#define SOC_DAPM_ENUM(xname, xenum) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = snd_soc_info_enum_double, \
@ -199,6 +215,7 @@ void snd_soc_dapm_free(struct snd_soc_device *socdev);
/* dapm events */
int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream,
int event);
int snd_soc_dapm_device_event(struct snd_soc_device *socdev, int event);
/* dapm sys fs - used by the core */
int snd_soc_dapm_sys_add(struct device *dev);
@ -272,7 +289,7 @@ struct snd_soc_dapm_widget {
/* external events */
unsigned short event_flags; /* flags to specify event types */
int (*event)(struct snd_soc_dapm_widget*, int);
int (*event)(struct snd_soc_dapm_widget*, struct snd_kcontrol *, int);
/* kcontrols that relate to this widget */
int num_kcontrols;

67
include/sound/soc.h
View File

@ -16,38 +16,63 @@
#include <linux/platform_device.h>
#include <linux/types.h>
#include <linux/workqueue.h>
#include <sound/driver.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/control.h>
#include <sound/ac97_codec.h>
#define SND_SOC_VERSION "0.13.1"
#define SND_SOC_VERSION "0.13.2"
/*
* Convenience kcontrol builders
*/
#define SOC_SINGLE_VALUE(reg,shift,mask,invert) ((reg) | ((shift) << 8) |\
((shift) << 12) | ((mask) << 16) | ((invert) << 24))
#define SOC_SINGLE_VALUE_EXT(reg,mask,invert) ((reg) | ((mask) << 16) |\
#define SOC_SINGLE_VALUE(reg, shift, max, invert) ((reg) | ((shift) << 8) |\
((shift) << 12) | ((max) << 16) | ((invert) << 24))
#define SOC_SINGLE_VALUE_EXT(reg, max, invert) ((reg) | ((max) << 16) |\
((invert) << 31))
#define SOC_SINGLE(xname, reg, shift, mask, invert) \
#define SOC_SINGLE(xname, reg, shift, max, invert) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\
.put = snd_soc_put_volsw, \
.private_value = SOC_SINGLE_VALUE(reg, shift, mask, invert) }
#define SOC_DOUBLE(xname, reg, shift_left, shift_right, mask, invert) \
.private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) }
#define SOC_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
SNDRV_CTL_ELEM_ACCESS_READWRITE,\
.tlv.p = (tlv_array), \
.info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\
.put = snd_soc_put_volsw, \
.private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) }
#define SOC_DOUBLE(xname, reg, shift_left, shift_right, max, invert) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
.info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \
.put = snd_soc_put_volsw, \
.private_value = (reg) | ((shift_left) << 8) | \
((shift_right) << 12) | ((mask) << 16) | ((invert) << 24) }
#define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, mask, invert) \
((shift_right) << 12) | ((max) << 16) | ((invert) << 24) }
#define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, max, invert) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
.info = snd_soc_info_volsw_2r, \
.get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \
.private_value = (reg_left) | ((shift) << 8) | \
((mask) << 12) | ((invert) << 20) | ((reg_right) << 24) }
((max) << 12) | ((invert) << 20) | ((reg_right) << 24) }
#define SOC_DOUBLE_TLV(xname, reg, shift_left, shift_right, max, invert, tlv_array) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
SNDRV_CTL_ELEM_ACCESS_READWRITE,\
.tlv.p = (tlv_array), \
.info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \
.put = snd_soc_put_volsw, \
.private_value = (reg) | ((shift_left) << 8) | \
((shift_right) << 12) | ((max) << 16) | ((invert) << 24) }
#define SOC_DOUBLE_R_TLV(xname, reg_left, reg_right, shift, max, invert, tlv_array) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
SNDRV_CTL_ELEM_ACCESS_READWRITE,\
.tlv.p = (tlv_array), \
.info = snd_soc_info_volsw_2r, \
.get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \
.private_value = (reg_left) | ((shift) << 8) | \
((max) << 12) | ((invert) << 20) | ((reg_right) << 24) }
#define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts) \
{ .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \
.mask = xmask, .texts = xtexts }
@ -104,10 +129,22 @@
#define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */
#define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated when not Tx/Rx */
/*
* DAI Sync
* Synchronous LR (Left Right) clocks and Frame signals.
*/
#define SND_SOC_DAIFMT_SYNC (0 << 5) /* Tx FRM = Rx FRM */
#define SND_SOC_DAIFMT_ASYNC (1 << 5) /* Tx FRM ~ Rx FRM */
/*
* TDM
*/
#define SND_SOC_DAIFMT_TDM (1 << 6)
/*
* DAI hardware signal inversions
*/
#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */
#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bclk + frm */
#define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal bclk + inv frm */
#define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert bclk + nor frm */
#define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert bclk + frm */
@ -410,6 +447,9 @@ struct snd_soc_dai_link {
/* codec/machine specific init - e.g. add machine controls */
int (*init)(struct snd_soc_codec *codec);
/* DAI pcm */
struct snd_pcm *pcm;
};
/* SoC machine */
@ -426,6 +466,9 @@ struct snd_soc_machine {
int (*resume_pre)(struct platform_device *pdev);
int (*resume_post)(struct platform_device *pdev);
/* callbacks */
int (*dapm_event)(struct snd_soc_machine *, int event);
/* CPU <--> Codec DAI links */
struct snd_soc_dai_link *dai_link;
int num_links;

1
include/sound/tea575x-tuner.h
View File

@ -30,6 +30,7 @@ struct snd_tea575x;
struct snd_tea575x_ops {
void (*write)(struct snd_tea575x *tea, unsigned int val);
unsigned int (*read)(struct snd_tea575x *tea);
void (*mute)(struct snd_tea575x *tea, unsigned int mute);
};
struct snd_tea575x {

22
include/sound/trident.h
View File

@ -26,19 +26,12 @@
#include "pcm.h"
#include "mpu401.h"
#include "ac97_codec.h"
#include "seq_midi_emul.h"
#include "seq_device.h"
#include "util_mem.h"
//#include "ainstr_iw.h"
//#include "ainstr_gf1.h"
#include "ainstr_simple.h"
#define TRIDENT_DEVICE_ID_DX ((PCI_VENDOR_ID_TRIDENT<<16)|PCI_DEVICE_ID_TRIDENT_4DWAVE_DX)
#define TRIDENT_DEVICE_ID_NX ((PCI_VENDOR_ID_TRIDENT<<16)|PCI_DEVICE_ID_TRIDENT_4DWAVE_NX)
#define TRIDENT_DEVICE_ID_SI7018 ((PCI_VENDOR_ID_SI<<16)|PCI_DEVICE_ID_SI_7018)
#define SNDRV_SEQ_DEV_ID_TRIDENT "trident-synth"
#define SNDRV_TRIDENT_VOICE_TYPE_PCM 0
#define SNDRV_TRIDENT_VOICE_TYPE_SYNTH 1
#define SNDRV_TRIDENT_VOICE_TYPE_MIDI 2
@ -257,16 +250,6 @@ struct snd_trident;
struct snd_trident_voice;
struct snd_trident_pcm_mixer;
struct snd_trident_sample_ops {
void (*sample_start)(struct snd_trident *gus, struct snd_trident_voice *voice, snd_seq_position_t position);
void (*sample_stop)(struct snd_trident *gus, struct snd_trident_voice *voice, int mode);
void (*sample_freq)(struct snd_trident *gus, struct snd_trident_voice *voice, snd_seq_frequency_t freq);
void (*sample_volume)(struct snd_trident *gus, struct snd_trident_voice *voice, struct snd_seq_ev_volume *volume);
void (*sample_loop)(struct snd_trident *card, struct snd_trident_voice *voice, struct snd_seq_ev_loop *loop);
void (*sample_pos)(struct snd_trident *card, struct snd_trident_voice *voice, snd_seq_position_t position);
void (*sample_private1)(struct snd_trident *card, struct snd_trident_voice *voice, unsigned char *data);
};
struct snd_trident_port {
struct snd_midi_channel_set * chset;
struct snd_trident * trident;
@ -300,7 +283,6 @@ struct snd_trident_voice {
unsigned char port;
unsigned char index;
struct snd_seq_instr instr;
struct snd_trident_sample_ops *sample_ops;
/* channel parameters */
@ -354,9 +336,6 @@ struct snd_4dwave {
int seq_client;
struct snd_trident_port seq_ports[4];
struct snd_simple_ops simple_ops;
struct snd_seq_kinstr_list *ilist;
struct snd_trident_voice voices[64];
int ChanSynthCount; /* number of allocated synth channels */
@ -416,7 +395,6 @@ struct snd_trident {
struct snd_pcm *foldback; /* Foldback PCM */
struct snd_pcm *spdif; /* SPDIF PCM */
struct snd_rawmidi *rmidi;
struct snd_seq_device *seq_dev;
struct snd_ac97_bus *ac97_bus;
struct snd_ac97 *ac97;

4
include/sound/version.h
View File

@ -1,3 +1,3 @@
/* include/version.h. Generated by alsa/ksync script. */
#define CONFIG_SND_VERSION "1.0.15"
#define CONFIG_SND_DATE " (Tue Nov 20 19:16:42 2007 UTC)"
#define CONFIG_SND_VERSION "1.0.16rc2"
#define CONFIG_SND_DATE " (Thu Jan 31 16:40:16 2008 UTC)"

2
sound/aoa/aoa.h
View File

@ -10,8 +10,6 @@
#define __AOA_H
#include <asm/prom.h>
#include <linux/module.h>
/* So apparently there's a reason for requiring driver.h to be included first! */
#include <sound/driver.h>
#include <sound/core.h>
#include <sound/asound.h>
#include <sound/control.h>

12
sound/aoa/codecs/snd-aoa-codec-onyx.c
View File

@ -138,6 +138,13 @@ static int onyx_snd_vol_put(struct snd_kcontrol *kcontrol,
struct onyx *onyx = snd_kcontrol_chip(kcontrol);
s8 l, r;
if (ucontrol->value.integer.value[0] < -128 + VOLUME_RANGE_SHIFT ||
ucontrol->value.integer.value[0] > -1 + VOLUME_RANGE_SHIFT)
return -EINVAL;
if (ucontrol->value.integer.value[1] < -128 + VOLUME_RANGE_SHIFT ||
ucontrol->value.integer.value[1] > -1 + VOLUME_RANGE_SHIFT)
return -EINVAL;
mutex_lock(&onyx->mutex);
onyx_read_register(onyx, ONYX_REG_DAC_ATTEN_LEFT, &l);
onyx_read_register(onyx, ONYX_REG_DAC_ATTEN_RIGHT, &r);
@ -206,6 +213,9 @@ static int onyx_snd_inputgain_put(struct snd_kcontrol *kcontrol,
struct onyx *onyx = snd_kcontrol_chip(kcontrol);
u8 v, n;
if (ucontrol->value.integer.value[0] < 3 + INPUTGAIN_RANGE_SHIFT ||
ucontrol->value.integer.value[0] > 28 + INPUTGAIN_RANGE_SHIFT)
return -EINVAL;
mutex_lock(&onyx->mutex);
onyx_read_register(onyx, ONYX_REG_ADC_CONTROL, &v);
n = v;
@ -272,6 +282,8 @@ static void onyx_set_capture_source(struct onyx *onyx, int mic)
static int onyx_snd_capture_source_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
if (ucontrol->value.enumerated.item[0] > 1)
return -EINVAL;
onyx_set_capture_source(snd_kcontrol_chip(kcontrol),
ucontrol->value.enumerated.item[0]);
return 1;

21
sound/aoa/codecs/snd-aoa-codec-tas.c
View File

@ -248,6 +248,13 @@ static int tas_snd_vol_put(struct snd_kcontrol *kcontrol,
{
struct tas *tas = snd_kcontrol_chip(kcontrol);
if (ucontrol->value.integer.value[0] < 0 ||
ucontrol->value.integer.value[0] > 177)
return -EINVAL;
if (ucontrol->value.integer.value[1] < 0 ||
ucontrol->value.integer.value[1] > 177)
return -EINVAL;
mutex_lock(&tas->mtx);
if (tas->cached_volume_l == ucontrol->value.integer.value[0]
&& tas->cached_volume_r == ucontrol->value.integer.value[1]) {
@ -401,6 +408,10 @@ static int tas_snd_drc_range_put(struct snd_kcontrol *kcontrol,
{
struct tas *tas = snd_kcontrol_chip(kcontrol);
if (ucontrol->value.integer.value[0] < 0 ||
ucontrol->value.integer.value[0] > TAS3004_DRC_MAX)
return -EINVAL;
mutex_lock(&tas->mtx);
if (tas->drc_range == ucontrol->value.integer.value[0]) {
mutex_unlock(&tas->mtx);
@ -447,7 +458,7 @@ static int tas_snd_drc_switch_put(struct snd_kcontrol *kcontrol,
return 0;
}
tas->drc_enabled = ucontrol->value.integer.value[0];
tas->drc_enabled = !!ucontrol->value.integer.value[0];
if (tas->hw_enabled)
tas3004_set_drc(tas);
mutex_unlock(&tas->mtx);
@ -494,6 +505,8 @@ static int tas_snd_capture_source_put(struct snd_kcontrol *kcontrol,
struct tas *tas = snd_kcontrol_chip(kcontrol);
int oldacr;
if (ucontrol->value.enumerated.item[0] > 1)
return -EINVAL;
mutex_lock(&tas->mtx);
oldacr = tas->acr;
@ -562,6 +575,9 @@ static int tas_snd_treble_put(struct snd_kcontrol *kcontrol,
{
struct tas *tas = snd_kcontrol_chip(kcontrol);
if (ucontrol->value.integer.value[0] < TAS3004_TREBLE_MIN ||
ucontrol->value.integer.value[0] > TAS3004_TREBLE_MAX)
return -EINVAL;
mutex_lock(&tas->mtx);
if (tas->treble == ucontrol->value.integer.value[0]) {
mutex_unlock(&tas->mtx);
@ -610,6 +626,9 @@ static int tas_snd_bass_put(struct snd_kcontrol *kcontrol,
{
struct tas *tas = snd_kcontrol_chip(kcontrol);
if (ucontrol->value.integer.value[0] < TAS3004_BASS_MIN ||
ucontrol->value.integer.value[0] > TAS3004_BASS_MAX)
return -EINVAL;
mutex_lock(&tas->mtx);
if (tas->bass == ucontrol->value.integer.value[0]) {
mutex_unlock(&tas->mtx);

2
sound/aoa/fabrics/snd-aoa-fabric-layout.c
View File

@ -600,7 +600,7 @@ static int n##_control_put(struct snd_kcontrol *kcontrol, \
struct gpio_runtime *gpio = snd_kcontrol_chip(kcontrol); \
if (gpio->methods && gpio->methods->get_##n) \
gpio->methods->set_##n(gpio, \
ucontrol->value.integer.value[0]); \
!!ucontrol->value.integer.value[0]); \
return 1; \
} \
static struct snd_kcontrol_new n##_ctl = { \

1
sound/aoa/soundbus/i2sbus/i2sbus-core.c
View File

@ -11,7 +11,6 @@
#include <linux/interrupt.h>
#include <linux/dma-mapping.h>
#include <sound/driver.h>
#include <sound/core.h>
#include <asm/macio.h>

10
sound/aoa/soundbus/i2sbus/i2sbus-pcm.c
View File

@ -8,9 +8,6 @@
#include <asm/io.h>
#include <linux/delay.h>
/* So apparently there's a reason for requiring driver.h
* to be included first, even if I don't know it... */
#include <sound/driver.h>
#include <sound/core.h>
#include <asm/macio.h>
#include <linux/pci.h>
@ -194,6 +191,12 @@ static int i2sbus_pcm_open(struct i2sbus_dev *i2sdev, int in)
hw->period_bytes_max = 16384;
hw->periods_min = 3;
hw->periods_max = MAX_DBDMA_COMMANDS;
err = snd_pcm_hw_constraint_integer(pi->substream->runtime,
SNDRV_PCM_HW_PARAM_PERIODS);
if (err < 0) {
result = err;
goto out_unlock;
}
list_for_each_entry(cii, &sdev->codec_list, list) {
if (cii->codec->open) {
err = cii->codec->open(cii, pi->substream);
@ -990,6 +993,7 @@ i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card,
if (dev->pcm->card != card) {
printk(KERN_ERR
"Can't attach same bus to different cards!\n");
err = -EINVAL;
goto out_put_ci_module;
}
err = snd_pcm_new_stream(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, 1);

1
sound/arm/aaci.c
View File

@ -23,7 +23,6 @@
#include <asm/irq.h>
#include <asm/sizes.h>
#include <sound/driver.h>
#include <sound/core.h>
#include <sound/initval.h>
#include <sound/ac97_codec.h>

1
sound/arm/devdma.c
View File

@ -12,7 +12,6 @@
#include <linux/device.h>
#include <linux/dma-mapping.h>
#include <sound/driver.h>
#include <sound/core.h>
#include <sound/pcm.h>

2
sound/arm/pxa2xx-ac97.c
View File

@ -18,7 +18,6 @@
#include <linux/wait.h>
#include <linux/delay.h>
#include <sound/driver.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/ac97_codec.h>
@ -352,6 +351,7 @@ static int __devinit pxa2xx_ac97_probe(struct platform_device *dev)
snprintf(card->longname, sizeof(card->longname),
"%s (%s)", dev->dev.driver->name, card->mixername);
snd_card_set_dev(card, &dev->dev);
ret = snd_card_register(card);
if (ret == 0) {
platform_set_drvdata(dev, card);

1
sound/arm/pxa2xx-pcm.c
View File

@ -16,7 +16,6 @@
#include <linux/slab.h>
#include <linux/dma-mapping.h>
#include <sound/driver.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>

1
sound/arm/sa11xx-uda1341.c
View File

@ -59,7 +59,6 @@
*
***************************************************************************************************/
#include <sound/driver.h>
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/init.h>

52
sound/core/control.c
View File

@ -19,7 +19,6 @@
*
*/
#include <sound/driver.h>
#include <linux/threads.h>
#include <linux/interrupt.h>
#include <linux/slab.h>
@ -232,8 +231,6 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol,
access = ncontrol->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE :
(ncontrol->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE|
SNDRV_CTL_ELEM_ACCESS_INACTIVE|
SNDRV_CTL_ELEM_ACCESS_DINDIRECT|
SNDRV_CTL_ELEM_ACCESS_INDIRECT|
SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE|
SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK));
kctl.info = ncontrol->info;
@ -692,7 +689,7 @@ int snd_ctl_elem_read(struct snd_card *card, struct snd_ctl_elem_value *control)
struct snd_kcontrol *kctl;
struct snd_kcontrol_volatile *vd;
unsigned int index_offset;
int result, indirect;
int result;
down_read(&card->controls_rwsem);
kctl = snd_ctl_find_id(card, &control->id);
@ -701,17 +698,12 @@ int snd_ctl_elem_read(struct snd_card *card, struct snd_ctl_elem_value *control)
} else {
index_offset = snd_ctl_get_ioff(kctl, &control->id);
vd = &kctl->vd[index_offset];
indirect = vd->access & SNDRV_CTL_ELEM_ACCESS_INDIRECT ? 1 : 0;
if (control->indirect != indirect) {
result = -EACCES;
} else {
if ((vd->access & SNDRV_CTL_ELEM_ACCESS_READ) && kctl->get != NULL) {
snd_ctl_build_ioff(&control->id, kctl, index_offset);
result = kctl->get(kctl, control);
} else {
result = -EPERM;
}
}
if ((vd->access & SNDRV_CTL_ELEM_ACCESS_READ) &&
kctl->get != NULL) {
snd_ctl_build_ioff(&control->id, kctl, index_offset);
result = kctl->get(kctl, control);
} else
result = -EPERM;
}
up_read(&card->controls_rwsem);
return result;
@ -748,7 +740,7 @@ int snd_ctl_elem_write(struct snd_card *card, struct snd_ctl_file *file,
struct snd_kcontrol *kctl;
struct snd_kcontrol_volatile *vd;
unsigned int index_offset;
int result, indirect;
int result;
down_read(&card->controls_rwsem);
kctl = snd_ctl_find_id(card, &control->id);
@ -757,23 +749,19 @@ int snd_ctl_elem_write(struct snd_card *card, struct snd_ctl_file *file,
} else {
index_offset = snd_ctl_get_ioff(kctl, &control->id);
vd = &kctl->vd[index_offset];
indirect = vd->access & SNDRV_CTL_ELEM_ACCESS_INDIRECT ? 1 : 0;
if (control->indirect != indirect) {
result = -EACCES;
if (!(vd->access & SNDRV_CTL_ELEM_ACCESS_WRITE) ||
kctl->put == NULL ||
(file && vd->owner && vd->owner != file)) {
result = -EPERM;
} else {
if (!(vd->access & SNDRV_CTL_ELEM_ACCESS_WRITE) ||
kctl->put == NULL ||
(file && vd->owner != NULL && vd->owner != file)) {
result = -EPERM;
} else {
snd_ctl_build_ioff(&control->id, kctl, index_offset);
result = kctl->put(kctl, control);
}
if (result > 0) {
up_read(&card->controls_rwsem);
snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &control->id);
return 0;
}
snd_ctl_build_ioff(&control->id, kctl, index_offset);
result = kctl->put(kctl, control);
}
if (result > 0) {
up_read(&card->controls_rwsem);
snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
&control->id);
return 0;
}
}
up_read(&card->controls_rwsem);

3
sound/core/control_compat.c
View File

@ -219,7 +219,8 @@ static int copy_ctl_value_from_user(struct snd_card *card,
struct snd_ctl_elem_value32 __user *data32,
int *typep, int *countp)
{
int i, type, count, size;
int i, type, size;
int uninitialized_var(count);
unsigned int indirect;
if (copy_from_user(&data->id, &data32->id, sizeof(data->id)))

1
sound/core/device.c
View File

@ -19,7 +19,6 @@
*
*/
#include <sound/driver.h>
#include <linux/slab.h>
#include <linux/time.h>
#include <linux/errno.h>

1
sound/core/hwdep.c
View File

@ -19,7 +19,6 @@
*
*/
#include <sound/driver.h>
#include <linux/major.h>
#include <linux/init.h>
#include <linux/slab.h>

1
sound/core/info.c
View File

@ -19,7 +19,6 @@
*
*/
#include <sound/driver.h>
#include <linux/init.h>
#include <linux/time.h>
#include <linux/smp_lock.h>

3
sound/core/info_oss.c
View File

@ -19,7 +19,6 @@
*
*/
#include <sound/driver.h>
#include <linux/slab.h>
#include <linux/time.h>
#include <linux/string.h>
@ -66,8 +65,6 @@ int snd_oss_info_register(int dev, int num, char *string)
EXPORT_SYMBOL(snd_oss_info_register);
extern void snd_card_info_read_oss(struct snd_info_buffer *buffer);
static int snd_sndstat_show_strings(struct snd_info_buffer *buf, char *id, int dev)
{
int idx, ok = -1;

39
sound/core/init.c
View File

@ -19,7 +19,6 @@
*
*/
#include <sound/driver.h>
#include <linux/init.h>
#include <linux/sched.h>
#include <linux/file.h>
@ -43,6 +42,40 @@ EXPORT_SYMBOL(snd_cards);
static DEFINE_MUTEX(snd_card_mutex);
static char *slots[SNDRV_CARDS];
module_param_array(slots, charp, NULL, 0444);
MODULE_PARM_DESC(slots, "Module names assigned to the slots.");
/* return non-zero if the given index is already reserved for another
* module via slots option
*/
static int module_slot_mismatch(struct module *module, int idx)
{
#ifdef MODULE
char *s1, *s2;
if (!module || !module->name || !slots[idx])
return 0;
s1 = slots[idx];
s2 = module->name;
/* compare module name strings
* hyphens are handled as equivalent with underscore
*/
for (;;) {
char c1 = *s1++;
char c2 = *s2++;
if (c1 == '-')
c1 = '_';
if (c2 == '-')
c2 = '_';
if (c1 != c2)
return 1;
if (!c1)
break;
}
#endif
return 0;
}
#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE)
int (*snd_mixer_oss_notify_callback)(struct snd_card *card, int free_flag);
EXPORT_SYMBOL(snd_mixer_oss_notify_callback);
@ -115,6 +148,8 @@ struct snd_card *snd_card_new(int idx, const char *xid,
for (idx2 = 0; idx2 < SNDRV_CARDS; idx2++)
/* idx == -1 == 0xffff means: take any free slot */
if (~snd_cards_lock & idx & 1<<idx2) {
if (module_slot_mismatch(module, idx2))
continue;
idx = idx2;
if (idx >= snd_ecards_limit)
snd_ecards_limit = idx + 1;
@ -304,8 +339,8 @@ int snd_card_disconnect(struct snd_card *card)
list_add(&mfile->shutdown_list, &shutdown_files);
spin_unlock(&shutdown_lock);
fops_get(&snd_shutdown_f_ops);
mfile->file->f_op = &snd_shutdown_f_ops;
fops_get(mfile->file->f_op);
mfile = mfile->next;
}

1
sound/core/isadma.c
View File

@ -26,7 +26,6 @@
#undef HAVE_REALLY_SLOW_DMA_CONTROLLER
#include <sound/driver.h>
#include <sound/core.h>
#include <asm/dma.h>

1
sound/core/memalloc.c
View File

@ -568,6 +568,7 @@ static ssize_t snd_mem_proc_write(struct file *file, const char __user * buffer,
if (pci_set_dma_mask(pci, mask) < 0 ||
pci_set_consistent_dma_mask(pci, mask) < 0) {
printk(KERN_ERR "snd-page-alloc: cannot set DMA mask %lx for pci %04x:%04x\n", mask, vendor, device);
pci_dev_put(pci);
return count;
}
}

2
sound/core/memory.c
View File

@ -20,9 +20,9 @@
*
*/
#include <linux/module.h>
#include <asm/io.h>
#include <asm/uaccess.h>
#include <sound/core.h>
/**
* copy_to_user_fromio - copy data from mmio-space to user-space

1
sound/core/misc.c
View File

@ -19,7 +19,6 @@
*
*/
#include <sound/driver.h>
#include <linux/init.h>
#include <linux/time.h>
#include <linux/ioport.h>

1
sound/core/oss/copy.c
View File

@ -19,7 +19,6 @@
*
*/
#include <sound/driver.h>
#include <linux/time.h>
#include <sound/core.h>
#include <sound/pcm.h>

1
sound/core/oss/io.c
View File

@ -19,7 +19,6 @@
*
*/
#include <sound/driver.h>
#include <linux/time.h>
#include <sound/core.h>
#include <sound/pcm.h>

1
sound/core/oss/linear.c
View File

@ -20,7 +20,6 @@
*
*/
#include <sound/driver.h>
#include <linux/time.h>
#include <sound/core.h>
#include <sound/pcm.h>

1
sound/core/oss/mixer_oss.c
View File

@ -19,7 +19,6 @@
*
*/
#include <sound/driver.h>
#include <linux/init.h>
#include <linux/slab.h>
#include <linux/time.h>

1
sound/core/oss/mulaw.c
View File

@ -21,7 +21,6 @@
*
*/
#include <sound/driver.h>
#include <linux/time.h>
#include <sound/core.h>
#include <sound/pcm.h>

4
sound/core/oss/pcm_oss.c
View File

@ -26,7 +26,6 @@
#define OSS_DEBUG
#endif
#include <sound/driver.h>
#include <linux/init.h>
#include <linux/slab.h>
#include <linux/time.h>
@ -985,10 +984,8 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream)
sw_params->stop_threshold = runtime->buffer_size;
sw_params->tstamp_mode = SNDRV_PCM_TSTAMP_NONE;
sw_params->period_step = 1;
sw_params->sleep_min = 0;
sw_params->avail_min = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
1 : runtime->period_size;
sw_params->xfer_align = 1;
if (atomic_read(&substream->mmap_count) ||
substream->oss.setup.nosilence) {
sw_params->silence_threshold = 0;
@ -1624,6 +1621,7 @@ static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file)
snd_pcm_format_set_silence(runtime->format,
runtime->oss.buffer,
size1);
size1 /= runtime->channels; /* frames */
fs = snd_enter_user();
snd_pcm_lib_write(substream, (void __user *)runtime->oss.buffer, size1);
snd_leave_user(fs);

1
sound/core/oss/pcm_plugin.c
View File

@ -24,7 +24,6 @@
#define PLUGIN_DEBUG
#endif
#include <sound/driver.h>
#include <linux/slab.h>
#include <linux/time.h>
#include <linux/vmalloc.h>

1
sound/core/oss/rate.c
View File

@ -19,7 +19,6 @@
*
*/
#include <sound/driver.h>
#include <linux/time.h>
#include <sound/core.h>
#include <sound/pcm.h>

1
sound/core/oss/route.c
View File

@ -19,7 +19,6 @@
*
*/
#include <sound/driver.h>
#include <linux/slab.h>
#include <linux/time.h>
#include <sound/core.h>

15
sound/core/pcm.c
View File

@ -19,7 +19,6 @@
*
*/
#include <sound/driver.h>
#include <linux/init.h>
#include <linux/slab.h>
#include <linux/time.h>
@ -228,7 +227,7 @@ static char *snd_pcm_subformat_names[] = {
static char *snd_pcm_tstamp_mode_names[] = {
TSTAMP(NONE),
TSTAMP(MMAP),
TSTAMP(ENABLE),
};
static const char *snd_pcm_stream_name(int stream)
@ -359,7 +358,6 @@ static void snd_pcm_substream_proc_hw_params_read(struct snd_info_entry *entry,
snd_iprintf(buffer, "rate: %u (%u/%u)\n", runtime->rate, runtime->rate_num, runtime->rate_den);
snd_iprintf(buffer, "period_size: %lu\n", runtime->period_size);
snd_iprintf(buffer, "buffer_size: %lu\n", runtime->buffer_size);
snd_iprintf(buffer, "tick_time: %u\n", runtime->tick_time);
#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE)
if (substream->oss.oss) {
snd_iprintf(buffer, "OSS format: %s\n", snd_pcm_oss_format_name(runtime->oss.format));
@ -387,9 +385,7 @@ static void snd_pcm_substream_proc_sw_params_read(struct snd_info_entry *entry,
}
snd_iprintf(buffer, "tstamp_mode: %s\n", snd_pcm_tstamp_mode_name(runtime->tstamp_mode));
snd_iprintf(buffer, "period_step: %u\n", runtime->period_step);
snd_iprintf(buffer, "sleep_min: %u\n", runtime->sleep_min);
snd_iprintf(buffer, "avail_min: %lu\n", runtime->control->avail_min);
snd_iprintf(buffer, "xfer_align: %lu\n", runtime->xfer_align);
snd_iprintf(buffer, "start_threshold: %lu\n", runtime->start_threshold);
snd_iprintf(buffer, "stop_threshold: %lu\n", runtime->stop_threshold);
snd_iprintf(buffer, "silence_threshold: %lu\n", runtime->silence_threshold);
@ -765,12 +761,6 @@ static int snd_pcm_dev_free(struct snd_device *device)
return snd_pcm_free(pcm);
}
static void snd_pcm_tick_timer_func(unsigned long data)
{
struct snd_pcm_substream *substream = (struct snd_pcm_substream *) data;
snd_pcm_tick_elapsed(substream);
}
int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream,
struct file *file,
struct snd_pcm_substream **rsubstream)
@ -877,9 +867,6 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream,
memset((void*)runtime->control, 0, size);
init_waitqueue_head(&runtime->sleep);
init_timer(&runtime->tick_timer);
runtime->tick_timer.function = snd_pcm_tick_timer_func;
runtime->tick_timer.data = (unsigned long) substream;
runtime->status->state = SNDRV_PCM_STATE_OPEN;

1
sound/core/pcm_compat.c
View File

@ -484,6 +484,7 @@ static long snd_pcm_ioctl_compat(struct file *file, unsigned int cmd, unsigned l
case SNDRV_PCM_IOCTL_PVERSION:
case SNDRV_PCM_IOCTL_INFO:
case SNDRV_PCM_IOCTL_TSTAMP:
case SNDRV_PCM_IOCTL_TTSTAMP:
case SNDRV_PCM_IOCTL_HWSYNC:
case SNDRV_PCM_IOCTL_PREPARE:
case SNDRV_PCM_IOCTL_RESET:

339
sound/core/pcm_lib.c
View File

@ -20,7 +20,6 @@
*
*/
#include <sound/driver.h>
#include <linux/slab.h>
#include <linux/time.h>
#include <sound/core.h>
@ -145,11 +144,11 @@ static inline snd_pcm_uframes_t snd_pcm_update_hw_ptr_pos(struct snd_pcm_substre
{
snd_pcm_uframes_t pos;
if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE)
snd_pcm_gettime(runtime, (struct timespec *)&runtime->status->tstamp);
pos = substream->ops->pointer(substream);
if (pos == SNDRV_PCM_POS_XRUN)
return pos; /* XRUN */
if (runtime->tstamp_mode & SNDRV_PCM_TSTAMP_MMAP)
getnstimeofday((struct timespec *)&runtime->status->tstamp);
#ifdef CONFIG_SND_DEBUG
if (pos >= runtime->buffer_size) {
snd_printk(KERN_ERR "BUG: stream = %i, pos = 0x%lx, buffer size = 0x%lx, period size = 0x%lx\n", substream->stream, pos, runtime->buffer_size, runtime->period_size);
@ -1139,7 +1138,7 @@ EXPORT_SYMBOL(snd_pcm_hw_constraint_step);
static int snd_pcm_hw_rule_pow2(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule)
{
static int pow2_sizes[] = {
static unsigned int pow2_sizes[] = {
1<<0, 1<<1, 1<<2, 1<<3, 1<<4, 1<<5, 1<<6, 1<<7,
1<<8, 1<<9, 1<<10, 1<<11, 1<<12, 1<<13, 1<<14, 1<<15,
1<<16, 1<<17, 1<<18, 1<<19, 1<<20, 1<<21, 1<<22, 1<<23,
@ -1451,108 +1450,13 @@ int snd_pcm_lib_ioctl(struct snd_pcm_substream *substream,
EXPORT_SYMBOL(snd_pcm_lib_ioctl);
/*
* Conditions
*/
static void snd_pcm_system_tick_set(struct snd_pcm_substream *substream,
unsigned long ticks)
{
struct snd_pcm_runtime *runtime = substream->runtime;
if (ticks == 0)
del_timer(&runtime->tick_timer);
else {
ticks += (1000000 / HZ) - 1;
ticks /= (1000000 / HZ);
mod_timer(&runtime->tick_timer, jiffies + ticks);
}
}
/* Temporary alias */
void snd_pcm_tick_set(struct snd_pcm_substream *substream, unsigned long ticks)
{
snd_pcm_system_tick_set(substream, ticks);
}
void snd_pcm_tick_prepare(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
snd_pcm_uframes_t frames = ULONG_MAX;
snd_pcm_uframes_t avail, dist;
unsigned int ticks;
u_int64_t n;
u_int32_t r;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
if (runtime->silence_size >= runtime->boundary) {
frames = 1;
} else if (runtime->silence_size > 0 &&
runtime->silence_filled < runtime->buffer_size) {
snd_pcm_sframes_t noise_dist;
noise_dist = snd_pcm_playback_hw_avail(runtime) + runtime->silence_filled;
if (noise_dist > (snd_pcm_sframes_t)runtime->silence_threshold)
frames = noise_dist - runtime->silence_threshold;
}
avail = snd_pcm_playback_avail(runtime);
} else {
avail = snd_pcm_capture_avail(runtime);
}
if (avail < runtime->control->avail_min) {
snd_pcm_sframes_t n = runtime->control->avail_min - avail;
if (n > 0 && frames > (snd_pcm_uframes_t)n)
frames = n;
}
if (avail < runtime->buffer_size) {
snd_pcm_sframes_t n = runtime->buffer_size - avail;
if (n > 0 && frames > (snd_pcm_uframes_t)n)
frames = n;
}
if (frames == ULONG_MAX) {
snd_pcm_tick_set(substream, 0);
return;
}
dist = runtime->status->hw_ptr - runtime->hw_ptr_base;
/* Distance to next interrupt */
dist = runtime->period_size - dist % runtime->period_size;
if (dist <= frames) {
snd_pcm_tick_set(substream, 0);
return;
}
/* the base time is us */
n = frames;
n *= 1000000;
div64_32(&n, runtime->tick_time * runtime->rate, &r);
ticks = n + (r > 0 ? 1 : 0);
if (ticks < runtime->sleep_min)
ticks = runtime->sleep_min;
snd_pcm_tick_set(substream, (unsigned long) ticks);
}
void snd_pcm_tick_elapsed(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime;
unsigned long flags;
snd_assert(substream != NULL, return);
runtime = substream->runtime;
snd_assert(runtime != NULL, return);
snd_pcm_stream_lock_irqsave(substream, flags);
if (!snd_pcm_running(substream) ||
snd_pcm_update_hw_ptr(substream) < 0)
goto _end;
if (runtime->sleep_min)
snd_pcm_tick_prepare(substream);
_end:
snd_pcm_stream_unlock_irqrestore(substream, flags);
}
/**
* snd_pcm_period_elapsed - update the pcm status for the next period
* @substream: the pcm substream instance
*
* This function is called from the interrupt handler when the
* PCM has processed the period size. It will update the current
* pointer, set up the tick, wake up sleepers, etc.
* pointer, wake up sleepers, etc.
*
* Even if more than one periods have elapsed since the last call, you
* have to call this only once.
@ -1576,8 +1480,6 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream)
if (substream->timer_running)
snd_timer_interrupt(substream->timer, 1);
if (runtime->sleep_min)
snd_pcm_tick_prepare(substream);
_end:
snd_pcm_stream_unlock_irqrestore(substream, flags);
if (runtime->transfer_ack_end)
@ -1587,6 +1489,71 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream)
EXPORT_SYMBOL(snd_pcm_period_elapsed);
/*
* Wait until avail_min data becomes available
* Returns a negative error code if any error occurs during operation.
* The available space is stored on availp. When err = 0 and avail = 0
* on the capture stream, it indicates the stream is in DRAINING state.
*/
static int wait_for_avail_min(struct snd_pcm_substream *substream,
snd_pcm_uframes_t *availp)
{
struct snd_pcm_runtime *runtime = substream->runtime;
int is_playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
wait_queue_t wait;
int err = 0;
snd_pcm_uframes_t avail = 0;
long tout;
init_waitqueue_entry(&wait, current);
add_wait_queue(&runtime->sleep, &wait);
for (;;) {
if (signal_pending(current)) {
err = -ERESTARTSYS;
break;
}
set_current_state(TASK_INTERRUPTIBLE);
snd_pcm_stream_unlock_irq(substream);
tout = schedule_timeout(msecs_to_jiffies(10000));
snd_pcm_stream_lock_irq(substream);
switch (runtime->status->state) {
case SNDRV_PCM_STATE_SUSPENDED:
err = -ESTRPIPE;
goto _endloop;
case SNDRV_PCM_STATE_XRUN:
err = -EPIPE;
goto _endloop;
case SNDRV_PCM_STATE_DRAINING:
if (is_playback)
err = -EPIPE;
else
avail = 0; /* indicate draining */
goto _endloop;
case SNDRV_PCM_STATE_OPEN:
case SNDRV_PCM_STATE_SETUP:
case SNDRV_PCM_STATE_DISCONNECTED:
err = -EBADFD;
goto _endloop;
}
if (!tout) {
snd_printd("%s write error (DMA or IRQ trouble?)\n",
is_playback ? "playback" : "capture");
err = -EIO;
break;
}
if (is_playback)
avail = snd_pcm_playback_avail(runtime);
else
avail = snd_pcm_capture_avail(runtime);
if (avail >= runtime->control->avail_min)
break;
}
_endloop:
remove_wait_queue(&runtime->sleep, &wait);
*availp = avail;
return err;
}
static int snd_pcm_lib_write_transfer(struct snd_pcm_substream *substream,
unsigned int hwoff,
unsigned long data, unsigned int off,
@ -1624,8 +1591,6 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream,
if (size == 0)
return 0;
if (size > runtime->xfer_align)
size -= size % runtime->xfer_align;
snd_pcm_stream_lock_irq(substream);
switch (runtime->status->state) {
@ -1648,84 +1613,18 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream,
snd_pcm_uframes_t frames, appl_ptr, appl_ofs;
snd_pcm_uframes_t avail;
snd_pcm_uframes_t cont;
if (runtime->sleep_min == 0 && runtime->status->state == SNDRV_PCM_STATE_RUNNING)
if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
snd_pcm_update_hw_ptr(substream);
avail = snd_pcm_playback_avail(runtime);
if (((avail < runtime->control->avail_min && size > avail) ||
(size >= runtime->xfer_align && avail < runtime->xfer_align))) {
wait_queue_t wait;
enum { READY, SIGNALED, ERROR, SUSPENDED, EXPIRED, DROPPED } state;
long tout;
if (!avail) {
if (nonblock) {
err = -EAGAIN;
goto _end_unlock;
}
init_waitqueue_entry(&wait, current);
add_wait_queue(&runtime->sleep, &wait);
while (1) {
if (signal_pending(current)) {
state = SIGNALED;
break;
}
set_current_state(TASK_INTERRUPTIBLE);
snd_pcm_stream_unlock_irq(substream);
tout = schedule_timeout(10 * HZ);
snd_pcm_stream_lock_irq(substream);
if (tout == 0) {
if (runtime->status->state != SNDRV_PCM_STATE_PREPARED &&
runtime->status->state != SNDRV_PCM_STATE_PAUSED) {
state = runtime->status->state == SNDRV_PCM_STATE_SUSPENDED ? SUSPENDED : EXPIRED;
break;
}
}
switch (runtime->status->state) {
case SNDRV_PCM_STATE_XRUN:
case SNDRV_PCM_STATE_DRAINING:
state = ERROR;
goto _end_loop;
case SNDRV_PCM_STATE_SUSPENDED:
state = SUSPENDED;
goto _end_loop;
case SNDRV_PCM_STATE_SETUP:
state = DROPPED;
goto _end_loop;
default:
break;
}
avail = snd_pcm_playback_avail(runtime);
if (avail >= runtime->control->avail_min) {
state = READY;
break;
}
}
_end_loop:
remove_wait_queue(&runtime->sleep, &wait);
switch (state) {
case ERROR:
err = -EPIPE;
err = wait_for_avail_min(substream, &avail);
if (err < 0)
goto _end_unlock;
case SUSPENDED:
err = -ESTRPIPE;
goto _end_unlock;
case SIGNALED:
err = -ERESTARTSYS;
goto _end_unlock;
case EXPIRED:
snd_printd("playback write error (DMA or IRQ trouble?)\n");
err = -EIO;
goto _end_unlock;
case DROPPED:
err = -EBADFD;
goto _end_unlock;
default:
break;
}
}
if (avail > runtime->xfer_align)
avail -= avail % runtime->xfer_align;
frames = size > avail ? avail : size;
cont = runtime->buffer_size - runtime->control->appl_ptr % runtime->buffer_size;
if (frames > cont)
@ -1763,9 +1662,6 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream,
if (err < 0)
goto _end_unlock;
}
if (runtime->sleep_min &&
runtime->status->state == SNDRV_PCM_STATE_RUNNING)
snd_pcm_tick_prepare(substream);
}
_end_unlock:
snd_pcm_stream_unlock_irq(substream);
@ -1893,8 +1789,6 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream,
if (size == 0)
return 0;
if (size > runtime->xfer_align)
size -= size % runtime->xfer_align;
snd_pcm_stream_lock_irq(substream);
switch (runtime->status->state) {
@ -1924,91 +1818,25 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream,
snd_pcm_uframes_t frames, appl_ptr, appl_ofs;
snd_pcm_uframes_t avail;
snd_pcm_uframes_t cont;
if (runtime->sleep_min == 0 && runtime->status->state == SNDRV_PCM_STATE_RUNNING)
if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
snd_pcm_update_hw_ptr(substream);
__draining:
avail = snd_pcm_capture_avail(runtime);
if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) {
if (avail < runtime->xfer_align) {
err = -EPIPE;
if (!avail) {
if (runtime->status->state ==
SNDRV_PCM_STATE_DRAINING) {
snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP);
goto _end_unlock;
}
} else if ((avail < runtime->control->avail_min && size > avail) ||
(size >= runtime->xfer_align && avail < runtime->xfer_align)) {
wait_queue_t wait;
enum { READY, SIGNALED, ERROR, SUSPENDED, EXPIRED, DROPPED } state;
long tout;
if (nonblock) {
err = -EAGAIN;
goto _end_unlock;
}
init_waitqueue_entry(&wait, current);
add_wait_queue(&runtime->sleep, &wait);
while (1) {
if (signal_pending(current)) {
state = SIGNALED;
break;
}
set_current_state(TASK_INTERRUPTIBLE);
snd_pcm_stream_unlock_irq(substream);
tout = schedule_timeout(10 * HZ);
snd_pcm_stream_lock_irq(substream);
if (tout == 0) {
if (runtime->status->state != SNDRV_PCM_STATE_PREPARED &&
runtime->status->state != SNDRV_PCM_STATE_PAUSED) {
state = runtime->status->state == SNDRV_PCM_STATE_SUSPENDED ? SUSPENDED : EXPIRED;
break;
}
}
switch (runtime->status->state) {
case SNDRV_PCM_STATE_XRUN:
state = ERROR;
goto _end_loop;
case SNDRV_PCM_STATE_SUSPENDED:
state = SUSPENDED;
goto _end_loop;
case SNDRV_PCM_STATE_DRAINING:
goto __draining;
case SNDRV_PCM_STATE_SETUP:
state = DROPPED;
goto _end_loop;
default:
break;
}
avail = snd_pcm_capture_avail(runtime);
if (avail >= runtime->control->avail_min) {
state = READY;
break;
}
}
_end_loop:
remove_wait_queue(&runtime->sleep, &wait);
switch (state) {
case ERROR:
err = -EPIPE;
err = wait_for_avail_min(substream, &avail);
if (err < 0)
goto _end_unlock;
case SUSPENDED:
err = -ESTRPIPE;
goto _end_unlock;
case SIGNALED:
err = -ERESTARTSYS;
goto _end_unlock;
case EXPIRED:
snd_printd("capture read error (DMA or IRQ trouble?)\n");
err = -EIO;
goto _end_unlock;
case DROPPED:
err = -EBADFD;
goto _end_unlock;
default:
break;
}
if (!avail)
continue; /* draining */
}
if (avail > runtime->xfer_align)
avail -= avail % runtime->xfer_align;
frames = size > avail ? avail : size;
cont = runtime->buffer_size - runtime->control->appl_ptr % runtime->buffer_size;
if (frames > cont)
@ -2040,9 +1868,6 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream,
offset += frames;
size -= frames;
xfer += frames;
if (runtime->sleep_min &&
runtime->status->state == SNDRV_PCM_STATE_RUNNING)
snd_pcm_tick_prepare(substream);
}
_end_unlock:
snd_pcm_stream_unlock_irq(substream);

1
sound/core/pcm_memory.c
View File

@ -19,7 +19,6 @@
*
*/
#include <sound/driver.h>
#include <asm/io.h>
#include <linux/time.h>
#include <linux/init.h>

3
sound/core/pcm_misc.c
View File

@ -19,7 +19,6 @@
*
*/
#include <sound/driver.h>
#include <linux/time.h>
#include <sound/core.h>
#include <sound/pcm.h>
@ -75,7 +74,7 @@ static struct pcm_format_data pcm_formats[SNDRV_PCM_FORMAT_LAST+1] = {
},
[SNDRV_PCM_FORMAT_U24_BE] = {
.width = 24, .phys = 32, .le = 0, .signd = 0,
.silence = { 0x80, 0x00, 0x00 },
.silence = { 0x00, 0x80, 0x00, 0x00 },
},
[SNDRV_PCM_FORMAT_S32_LE] = {
.width = 32, .phys = 32, .le = 1, .signd = 1,

141
sound/core/pcm_native.c
View File

@ -19,7 +19,6 @@
*
*/
#include <sound/driver.h>
#include <linux/mm.h>
#include <linux/file.h>
#include <linux/slab.h>
@ -413,7 +412,6 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream,
runtime->period_size = params_period_size(params);
runtime->periods = params_periods(params);
runtime->buffer_size = params_buffer_size(params);
runtime->tick_time = params_tick_time(params);
runtime->info = params->info;
runtime->rate_num = params->rate_num;
runtime->rate_den = params->rate_den;
@ -433,9 +431,7 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream,
/* Default sw params */
runtime->tstamp_mode = SNDRV_PCM_TSTAMP_NONE;
runtime->period_step = 1;
runtime->sleep_min = 0;
runtime->control->avail_min = runtime->period_size;
runtime->xfer_align = runtime->period_size;
runtime->start_threshold = 1;
runtime->stop_threshold = runtime->buffer_size;
runtime->silence_threshold = 0;
@ -532,9 +528,6 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream,
return -EINVAL;
if (params->avail_min == 0)
return -EINVAL;
if (params->xfer_align == 0 ||
params->xfer_align % runtime->min_align != 0)
return -EINVAL;
if (params->silence_size >= runtime->boundary) {
if (params->silence_threshold != 0)
return -EINVAL;
@ -546,20 +539,14 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream,
}
snd_pcm_stream_lock_irq(substream);
runtime->tstamp_mode = params->tstamp_mode;
runtime->sleep_min = params->sleep_min;
runtime->period_step = params->period_step;
runtime->control->avail_min = params->avail_min;
runtime->start_threshold = params->start_threshold;
runtime->stop_threshold = params->stop_threshold;
runtime->silence_threshold = params->silence_threshold;
runtime->silence_size = params->silence_size;
runtime->xfer_align = params->xfer_align;
params->boundary = runtime->boundary;
if (snd_pcm_running(substream)) {
if (runtime->sleep_min)
snd_pcm_tick_prepare(substream);
else
snd_pcm_tick_set(substream, 0);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
runtime->silence_size > 0)
snd_pcm_playback_silence(substream, ULONG_MAX);
@ -595,12 +582,13 @@ int snd_pcm_status(struct snd_pcm_substream *substream,
status->trigger_tstamp = runtime->trigger_tstamp;
if (snd_pcm_running(substream)) {
snd_pcm_update_hw_ptr(substream);
if (runtime->tstamp_mode & SNDRV_PCM_TSTAMP_MMAP)
if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) {
status->tstamp = runtime->status->tstamp;
else
getnstimeofday(&status->tstamp);
} else
getnstimeofday(&status->tstamp);
goto _tstamp_end;
}
}
snd_pcm_gettime(runtime, &status->tstamp);
_tstamp_end:
status->appl_ptr = runtime->control->appl_ptr;
status->hw_ptr = runtime->status->hw_ptr;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
@ -688,7 +676,7 @@ static void snd_pcm_trigger_tstamp(struct snd_pcm_substream *substream)
if (runtime->trigger_master == NULL)
return;
if (runtime->trigger_master == substream) {
getnstimeofday(&runtime->trigger_tstamp);
snd_pcm_gettime(runtime, &runtime->trigger_tstamp);
} else {
snd_pcm_trigger_tstamp(runtime->trigger_master);
runtime->trigger_tstamp = runtime->trigger_master->runtime->trigger_tstamp;
@ -875,8 +863,6 @@ static void snd_pcm_post_start(struct snd_pcm_substream *substream, int state)
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
runtime->silence_size > 0)
snd_pcm_playback_silence(substream, ULONG_MAX);
if (runtime->sleep_min)
snd_pcm_tick_prepare(substream);
if (substream->timer)
snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSTART,
&runtime->trigger_tstamp);
@ -930,7 +916,6 @@ static void snd_pcm_post_stop(struct snd_pcm_substream *substream, int state)
snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSTOP,
&runtime->trigger_tstamp);
runtime->status->state = state;
snd_pcm_tick_set(substream, 0);
}
wake_up(&runtime->sleep);
}
@ -1014,12 +999,9 @@ static void snd_pcm_post_pause(struct snd_pcm_substream *substream, int push)
snd_timer_notify(substream->timer,
SNDRV_TIMER_EVENT_MPAUSE,
&runtime->trigger_tstamp);
snd_pcm_tick_set(substream, 0);
wake_up(&runtime->sleep);
} else {
runtime->status->state = SNDRV_PCM_STATE_RUNNING;
if (runtime->sleep_min)
snd_pcm_tick_prepare(substream);
if (substream->timer)
snd_timer_notify(substream->timer,
SNDRV_TIMER_EVENT_MCONTINUE,
@ -1074,7 +1056,6 @@ static void snd_pcm_post_suspend(struct snd_pcm_substream *substream, int state)
&runtime->trigger_tstamp);
runtime->status->suspended_state = runtime->status->state;
runtime->status->state = SNDRV_PCM_STATE_SUSPENDED;
snd_pcm_tick_set(substream, 0);
wake_up(&runtime->sleep);
}
@ -1177,8 +1158,6 @@ static void snd_pcm_post_resume(struct snd_pcm_substream *substream, int state)
snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MRESUME,
&runtime->trigger_tstamp);
runtime->status->state = runtime->status->suspended_state;
if (runtime->sleep_min)
snd_pcm_tick_prepare(substream);
}
static struct action_ops snd_pcm_action_resume = {
@ -1395,10 +1374,10 @@ static int snd_pcm_do_drain_init(struct snd_pcm_substream *substream, int state)
} else {
/* stop running stream */
if (runtime->status->state == SNDRV_PCM_STATE_RUNNING) {
int state = snd_pcm_capture_avail(runtime) > 0 ?
int new_state = snd_pcm_capture_avail(runtime) > 0 ?
SNDRV_PCM_STATE_DRAINING : SNDRV_PCM_STATE_SETUP;
snd_pcm_do_stop(substream, state);
snd_pcm_post_stop(substream, state);
snd_pcm_do_stop(substream, new_state);
snd_pcm_post_stop(substream, new_state);
}
}
return 0;
@ -2007,8 +1986,6 @@ int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream)
}
/* FIXME: this belong to lowlevel */
snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_TICK_TIME,
1000000 / HZ, 1000000 / HZ);
snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE);
return 0;
@ -2244,15 +2221,10 @@ static snd_pcm_sframes_t snd_pcm_playback_rewind(struct snd_pcm_substream *subst
}
if (frames > (snd_pcm_uframes_t)hw_avail)
frames = hw_avail;
else
frames -= frames % runtime->xfer_align;
appl_ptr = runtime->control->appl_ptr - frames;
if (appl_ptr < 0)
appl_ptr += runtime->boundary;
runtime->control->appl_ptr = appl_ptr;
if (runtime->status->state == SNDRV_PCM_STATE_RUNNING &&
runtime->sleep_min)
snd_pcm_tick_prepare(substream);
ret = frames;
__end:
snd_pcm_stream_unlock_irq(substream);
@ -2294,15 +2266,10 @@ static snd_pcm_sframes_t snd_pcm_capture_rewind(struct snd_pcm_substream *substr
}
if (frames > (snd_pcm_uframes_t)hw_avail)
frames = hw_avail;
else
frames -= frames % runtime->xfer_align;
appl_ptr = runtime->control->appl_ptr - frames;
if (appl_ptr < 0)
appl_ptr += runtime->boundary;
runtime->control->appl_ptr = appl_ptr;
if (runtime->status->state == SNDRV_PCM_STATE_RUNNING &&
runtime->sleep_min)
snd_pcm_tick_prepare(substream);
ret = frames;
__end:
snd_pcm_stream_unlock_irq(substream);
@ -2345,15 +2312,10 @@ static snd_pcm_sframes_t snd_pcm_playback_forward(struct snd_pcm_substream *subs
}
if (frames > (snd_pcm_uframes_t)avail)
frames = avail;
else
frames -= frames % runtime->xfer_align;
appl_ptr = runtime->control->appl_ptr + frames;
if (appl_ptr >= (snd_pcm_sframes_t)runtime->boundary)
appl_ptr -= runtime->boundary;
runtime->control->appl_ptr = appl_ptr;
if (runtime->status->state == SNDRV_PCM_STATE_RUNNING &&
runtime->sleep_min)
snd_pcm_tick_prepare(substream);
ret = frames;
__end:
snd_pcm_stream_unlock_irq(substream);
@ -2396,15 +2358,10 @@ static snd_pcm_sframes_t snd_pcm_capture_forward(struct snd_pcm_substream *subst
}
if (frames > (snd_pcm_uframes_t)avail)
frames = avail;
else
frames -= frames % runtime->xfer_align;
appl_ptr = runtime->control->appl_ptr + frames;
if (appl_ptr >= (snd_pcm_sframes_t)runtime->boundary)
appl_ptr -= runtime->boundary;
runtime->control->appl_ptr = appl_ptr;
if (runtime->status->state == SNDRV_PCM_STATE_RUNNING &&
runtime->sleep_min)
snd_pcm_tick_prepare(substream);
ret = frames;
__end:
snd_pcm_stream_unlock_irq(substream);
@ -2519,6 +2476,21 @@ static int snd_pcm_sync_ptr(struct snd_pcm_substream *substream,
return -EFAULT;
return 0;
}
static int snd_pcm_tstamp(struct snd_pcm_substream *substream, int __user *_arg)
{
struct snd_pcm_runtime *runtime = substream->runtime;
int arg;
if (get_user(arg, _arg))
return -EFAULT;
if (arg < 0 || arg > SNDRV_PCM_TSTAMP_TYPE_LAST)
return -EINVAL;
runtime->tstamp_type = SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY;
if (arg == SNDRV_PCM_TSTAMP_TYPE_MONOTONIC)
runtime->tstamp_type = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC;
return 0;
}
static int snd_pcm_common_ioctl1(struct file *file,
struct snd_pcm_substream *substream,
@ -2531,8 +2503,10 @@ static int snd_pcm_common_ioctl1(struct file *file,
return put_user(SNDRV_PCM_VERSION, (int __user *)arg) ? -EFAULT : 0;
case SNDRV_PCM_IOCTL_INFO:
return snd_pcm_info_user(substream, arg);
case SNDRV_PCM_IOCTL_TSTAMP: /* just for compatibility */
case SNDRV_PCM_IOCTL_TSTAMP: /* just for compatibility */
return 0;
case SNDRV_PCM_IOCTL_TTSTAMP:
return snd_pcm_tstamp(substream, arg);
case SNDRV_PCM_IOCTL_HW_REFINE:
return snd_pcm_hw_refine_user(substream, arg);
case SNDRV_PCM_IOCTL_HW_PARAMS:
@ -3018,26 +2992,23 @@ static unsigned int snd_pcm_capture_poll(struct file *file, poll_table * wait)
/*
* mmap status record
*/
static struct page * snd_pcm_mmap_status_nopage(struct vm_area_struct *area,
unsigned long address, int *type)
static int snd_pcm_mmap_status_fault(struct vm_area_struct *area,
struct vm_fault *vmf)
{
struct snd_pcm_substream *substream = area->vm_private_data;
struct snd_pcm_runtime *runtime;
struct page * page;
if (substream == NULL)
return NOPAGE_SIGBUS;
return VM_FAULT_SIGBUS;
runtime = substream->runtime;
page = virt_to_page(runtime->status);
get_page(page);
if (type)
*type = VM_FAULT_MINOR;
return page;
vmf->page = virt_to_page(runtime->status);
get_page(vmf->page);
return 0;
}
static struct vm_operations_struct snd_pcm_vm_ops_status =
{
.nopage = snd_pcm_mmap_status_nopage,
.fault = snd_pcm_mmap_status_fault,
};
static int snd_pcm_mmap_status(struct snd_pcm_substream *substream, struct file *file,
@ -3061,26 +3032,23 @@ static int snd_pcm_mmap_status(struct snd_pcm_substream *substream, struct file
/*
* mmap control record
*/
static struct page * snd_pcm_mmap_control_nopage(struct vm_area_struct *area,
unsigned long address, int *type)
static int snd_pcm_mmap_control_fault(struct vm_area_struct *area,
struct vm_fault *vmf)
{
struct snd_pcm_substream *substream = area->vm_private_data;
struct snd_pcm_runtime *runtime;
struct page * page;
if (substream == NULL)
return NOPAGE_SIGBUS;
return VM_FAULT_SIGBUS;
runtime = substream->runtime;
page = virt_to_page(runtime->control);
get_page(page);
if (type)
*type = VM_FAULT_MINOR;
return page;
vmf->page = virt_to_page(runtime->control);
get_page(vmf->page);
return 0;
}
static struct vm_operations_struct snd_pcm_vm_ops_control =
{
.nopage = snd_pcm_mmap_control_nopage,
.fault = snd_pcm_mmap_control_fault,
};
static int snd_pcm_mmap_control(struct snd_pcm_substream *substream, struct file *file,
@ -3117,10 +3085,10 @@ static int snd_pcm_mmap_control(struct snd_pcm_substream *substream, struct file
#endif /* coherent mmap */
/*
* nopage callback for mmapping a RAM page
* fault callback for mmapping a RAM page
*/
static struct page *snd_pcm_mmap_data_nopage(struct vm_area_struct *area,
unsigned long address, int *type)
static int snd_pcm_mmap_data_fault(struct vm_area_struct *area,
struct vm_fault *vmf)
{
struct snd_pcm_substream *substream = area->vm_private_data;
struct snd_pcm_runtime *runtime;
@ -3130,33 +3098,30 @@ static struct page *snd_pcm_mmap_data_nopage(struct vm_area_struct *area,
size_t dma_bytes;
if (substream == NULL)
return NOPAGE_SIGBUS;
return VM_FAULT_SIGBUS;
runtime = substream->runtime;
offset = area->vm_pgoff << PAGE_SHIFT;
offset += address - area->vm_start;
snd_assert((offset % PAGE_SIZE) == 0, return NOPAGE_SIGBUS);
offset = vmf->pgoff << PAGE_SHIFT;
dma_bytes = PAGE_ALIGN(runtime->dma_bytes);
if (offset > dma_bytes - PAGE_SIZE)
return NOPAGE_SIGBUS;
return VM_FAULT_SIGBUS;
if (substream->ops->page) {
page = substream->ops->page(substream, offset);
if (! page)
return NOPAGE_OOM; /* XXX: is this really due to OOM? */
if (!page)
return VM_FAULT_SIGBUS;
} else {
vaddr = runtime->dma_area + offset;
page = virt_to_page(vaddr);
}
get_page(page);
if (type)
*type = VM_FAULT_MINOR;
return page;
vmf->page = page;
return 0;
}
static struct vm_operations_struct snd_pcm_vm_ops_data =
{
.open = snd_pcm_mmap_data_open,
.close = snd_pcm_mmap_data_close,
.nopage = snd_pcm_mmap_data_nopage,
.fault = snd_pcm_mmap_data_fault,
};
/*

1
sound/core/pcm_timer.c
View File

@ -19,7 +19,6 @@
*
*/
#include <sound/driver.h>
#include <linux/time.h>
#include <sound/core.h>
#include <sound/pcm.h>

41
sound/core/rawmidi.c
View File

@ -19,7 +19,6 @@
*
*/
#include <sound/driver.h>
#include <sound/core.h>
#include <linux/major.h>
#include <linux/init.h>
@ -912,7 +911,8 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream,
}
static long snd_rawmidi_kernel_read1(struct snd_rawmidi_substream *substream,
unsigned char *buf, long count, int kernel)
unsigned char __user *userbuf,
unsigned char *kernelbuf, long count)
{
unsigned long flags;
long result = 0, count1;
@ -925,11 +925,11 @@ static long snd_rawmidi_kernel_read1(struct snd_rawmidi_substream *substream,
spin_lock_irqsave(&runtime->lock, flags);
if (count1 > (int)runtime->avail)
count1 = runtime->avail;
if (kernel) {
memcpy(buf + result, runtime->buffer + runtime->appl_ptr, count1);
} else {
if (kernelbuf)
memcpy(kernelbuf + result, runtime->buffer + runtime->appl_ptr, count1);
if (userbuf) {
spin_unlock_irqrestore(&runtime->lock, flags);
if (copy_to_user((char __user *)buf + result,
if (copy_to_user(userbuf + result,
runtime->buffer + runtime->appl_ptr, count1)) {
return result > 0 ? result : -EFAULT;
}
@ -949,7 +949,7 @@ long snd_rawmidi_kernel_read(struct snd_rawmidi_substream *substream,
unsigned char *buf, long count)
{
snd_rawmidi_input_trigger(substream, 1);
return snd_rawmidi_kernel_read1(substream, buf, count, 1);
return snd_rawmidi_kernel_read1(substream, NULL/*userbuf*/, buf, count);
}
static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t count,
@ -990,8 +990,9 @@ static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t coun
}
spin_unlock_irq(&runtime->lock);
count1 = snd_rawmidi_kernel_read1(substream,
(unsigned char __force *)buf,
count, 0);
(unsigned char __user *)buf,
NULL/*kernelbuf*/,
count);
if (count1 < 0)
return result > 0 ? result : count1;
result += count1;
@ -1132,13 +1133,15 @@ int snd_rawmidi_transmit(struct snd_rawmidi_substream *substream,
}
static long snd_rawmidi_kernel_write1(struct snd_rawmidi_substream *substream,
const unsigned char *buf, long count, int kernel)
const unsigned char __user *userbuf,
const unsigned char *kernelbuf,
long count)
{
unsigned long flags;
long count1, result;
struct snd_rawmidi_runtime *runtime = substream->runtime;
snd_assert(buf != NULL, return -EINVAL);
snd_assert(kernelbuf != NULL || userbuf != NULL, return -EINVAL);
snd_assert(runtime->buffer != NULL, return -EINVAL);
result = 0;
@ -1155,12 +1158,13 @@ static long snd_rawmidi_kernel_write1(struct snd_rawmidi_substream *substream,
count1 = count;
if (count1 > (long)runtime->avail)
count1 = runtime->avail;
if (kernel) {
memcpy(runtime->buffer + runtime->appl_ptr, buf, count1);
} else {
if (kernelbuf)
memcpy(runtime->buffer + runtime->appl_ptr,
kernelbuf + result, count1);
else if (userbuf) {
spin_unlock_irqrestore(&runtime->lock, flags);
if (copy_from_user(runtime->buffer + runtime->appl_ptr,
(char __user *)buf, count1)) {
userbuf + result, count1)) {
spin_lock_irqsave(&runtime->lock, flags);
result = result > 0 ? result : -EFAULT;
goto __end;
@ -1171,7 +1175,6 @@ static long snd_rawmidi_kernel_write1(struct snd_rawmidi_substream *substream,
runtime->appl_ptr %= runtime->buffer_size;
runtime->avail -= count1;
result += count1;
buf += count1;
count -= count1;
}
__end:
@ -1185,7 +1188,7 @@ static long snd_rawmidi_kernel_write1(struct snd_rawmidi_substream *substream,
long snd_rawmidi_kernel_write(struct snd_rawmidi_substream *substream,
const unsigned char *buf, long count)
{
return snd_rawmidi_kernel_write1(substream, buf, count, 1);
return snd_rawmidi_kernel_write1(substream, NULL, buf, count);
}
static ssize_t snd_rawmidi_write(struct file *file, const char __user *buf,
@ -1225,9 +1228,7 @@ static ssize_t snd_rawmidi_write(struct file *file, const char __user *buf,
spin_lock_irq(&runtime->lock);
}
spin_unlock_irq(&runtime->lock);
count1 = snd_rawmidi_kernel_write1(substream,
(unsigned char __force *)buf,
count, 0);
count1 = snd_rawmidi_kernel_write1(substream, buf, NULL, count);
if (count1 < 0)
return result > 0 ? result : count1;
result += count1;

1
sound/core/rtctimer.c
View File

@ -20,7 +20,6 @@
*
*/
#include <sound/driver.h>
#include <linux/init.h>
#include <linux/interrupt.h>
#include <linux/moduleparam.h>

8
sound/core/seq/Makefile
View File

@ -3,7 +3,6 @@
# Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
#
obj-$(CONFIG_SND) += instr/
ifeq ($(CONFIG_SND_SEQUENCER_OSS),y)
obj-$(CONFIG_SND_SEQUENCER) += oss/
endif
@ -15,7 +14,6 @@ snd-seq-objs := seq.o seq_lock.o seq_clientmgr.o seq_memory.o seq_queue.o \
snd-seq-midi-objs := seq_midi.o
snd-seq-midi-emul-objs := seq_midi_emul.o
snd-seq-midi-event-objs := seq_midi_event.o
snd-seq-instr-objs := seq_instr.o
snd-seq-dummy-objs := seq_dummy.o
snd-seq-virmidi-objs := seq_virmidi.o
@ -36,9 +34,7 @@ obj-$(CONFIG_SND_SEQ_DUMMY) += snd-seq-dummy.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_VIRMIDI) += snd-seq-virmidi.o snd-seq-midi-event.o
obj-$(call sequencer,$(CONFIG_SND_RAWMIDI)) += snd-seq-midi.o snd-seq-midi-event.o
obj-$(call sequencer,$(CONFIG_SND_OPL3_LIB)) += snd-seq-midi-event.o snd-seq-midi-emul.o snd-seq-instr.o
obj-$(call sequencer,$(CONFIG_SND_OPL4_LIB)) += snd-seq-midi-event.o snd-seq-midi-emul.o snd-seq-instr.o
obj-$(call sequencer,$(CONFIG_SND_GUS_SYNTH)) += snd-seq-midi-emul.o snd-seq-instr.o
obj-$(call sequencer,$(CONFIG_SND_OPL3_LIB)) += snd-seq-midi-event.o snd-seq-midi-emul.o
obj-$(call sequencer,$(CONFIG_SND_OPL4_LIB)) += snd-seq-midi-event.o snd-seq-midi-emul.o
obj-$(call sequencer,$(CONFIG_SND_SBAWE)) += snd-seq-midi-emul.o snd-seq-virmidi.o
obj-$(call sequencer,$(CONFIG_SND_EMU10K1)) += snd-seq-midi-emul.o snd-seq-virmidi.o
obj-$(call sequencer,$(CONFIG_SND_TRIDENT)) += snd-seq-midi-emul.o snd-seq-instr.o

23
sound/core/seq/instr/Makefile
View File

@ -1,23 +0,0 @@
#
# Makefile for ALSA
# Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
#
snd-ainstr-fm-objs := ainstr_fm.o
snd-ainstr-simple-objs := ainstr_simple.o
snd-ainstr-gf1-objs := ainstr_gf1.o
snd-ainstr-iw-objs := ainstr_iw.o
#
# this function returns:
# "m" - CONFIG_SND_SEQUENCER is m
# <empty string> - CONFIG_SND_SEQUENCER is undefined
# otherwise parameter #1 value
#
sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1)))
# Toplevel Module Dependency
obj-$(call sequencer,$(CONFIG_SND_OPL3_LIB)) += snd-ainstr-fm.o
obj-$(call sequencer,$(CONFIG_SND_OPL4_LIB)) += snd-ainstr-fm.o
obj-$(call sequencer,$(CONFIG_SND_GUS_SYNTH)) += snd-ainstr-gf1.o snd-ainstr-simple.o snd-ainstr-iw.o
obj-$(call sequencer,$(CONFIG_SND_TRIDENT)) += snd-ainstr-simple.o

155
sound/core/seq/instr/ainstr_fm.c
View File

@ -1,155 +0,0 @@
/*
* FM (OPL2/3) Instrument routines
* Copyright (c) 2000 Uros Bizjak <uros@kss-loka.si>
*
* 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/init.h>
#include <sound/core.h>
#include <sound/ainstr_fm.h>
#include <sound/initval.h>
#include <asm/uaccess.h>
MODULE_AUTHOR("Uros Bizjak <uros@kss-loka.si>");
MODULE_DESCRIPTION("Advanced Linux Sound Architecture FM Instrument support.");
MODULE_LICENSE("GPL");
static int snd_seq_fm_put(void *private_data, struct snd_seq_kinstr *instr,
char __user *instr_data, long len, int atomic, int cmd)
{
struct fm_instrument *ip;
struct fm_xinstrument ix;
int idx;
if (cmd != SNDRV_SEQ_INSTR_PUT_CMD_CREATE)
return -EINVAL;
/* copy instrument data */
if (len < (long)sizeof(ix))
return -EINVAL;
if (copy_from_user(&ix, instr_data, sizeof(ix)))
return -EFAULT;
if (ix.stype != FM_STRU_INSTR)
return -EINVAL;
ip = (struct fm_instrument *)KINSTR_DATA(instr);
ip->share_id[0] = le32_to_cpu(ix.share_id[0]);
ip->share_id[1] = le32_to_cpu(ix.share_id[1]);
ip->share_id[2] = le32_to_cpu(ix.share_id[2]);
ip->share_id[3] = le32_to_cpu(ix.share_id[3]);
ip->type = ix.type;
for (idx = 0; idx < 4; idx++) {
ip->op[idx].am_vib = ix.op[idx].am_vib;
ip->op[idx].ksl_level = ix.op[idx].ksl_level;
ip->op[idx].attack_decay = ix.op[idx].attack_decay;
ip->op[idx].sustain_release = ix.op[idx].sustain_release;
ip->op[idx].wave_select = ix.op[idx].wave_select;
}
for (idx = 0; idx < 2; idx++) {
ip->feedback_connection[idx] = ix.feedback_connection[idx];
}
ip->echo_delay = ix.echo_delay;
ip->echo_atten = ix.echo_atten;
ip->chorus_spread = ix.chorus_spread;
ip->trnsps = ix.trnsps;
ip->fix_dur = ix.fix_dur;
ip->modes = ix.modes;
ip->fix_key = ix.fix_key;
return 0;
}
static int snd_seq_fm_get(void *private_data, struct snd_seq_kinstr *instr,
char __user *instr_data, long len, int atomic,
int cmd)
{
struct fm_instrument *ip;
struct fm_xinstrument ix;
int idx;
if (cmd != SNDRV_SEQ_INSTR_GET_CMD_FULL)
return -EINVAL;
if (len < (long)sizeof(ix))
return -ENOMEM;
memset(&ix, 0, sizeof(ix));
ip = (struct fm_instrument *)KINSTR_DATA(instr);
ix.stype = FM_STRU_INSTR;
ix.share_id[0] = cpu_to_le32(ip->share_id[0]);
ix.share_id[1] = cpu_to_le32(ip->share_id[1]);
ix.share_id[2] = cpu_to_le32(ip->share_id[2]);
ix.share_id[3] = cpu_to_le32(ip->share_id[3]);
ix.type = ip->type;
for (idx = 0; idx < 4; idx++) {
ix.op[idx].am_vib = ip->op[idx].am_vib;
ix.op[idx].ksl_level = ip->op[idx].ksl_level;
ix.op[idx].attack_decay = ip->op[idx].attack_decay;
ix.op[idx].sustain_release = ip->op[idx].sustain_release;
ix.op[idx].wave_select = ip->op[idx].wave_select;
}
for (idx = 0; idx < 2; idx++) {
ix.feedback_connection[idx] = ip->feedback_connection[idx];
}
if (copy_to_user(instr_data, &ix, sizeof(ix)))
return -EFAULT;
ix.echo_delay = ip->echo_delay;
ix.echo_atten = ip->echo_atten;
ix.chorus_spread = ip->chorus_spread;
ix.trnsps = ip->trnsps;
ix.fix_dur = ip->fix_dur;
ix.modes = ip->modes;
ix.fix_key = ip->fix_key;
return 0;
}
static int snd_seq_fm_get_size(void *private_data, struct snd_seq_kinstr *instr,
long *size)
{
*size = sizeof(struct fm_xinstrument);
return 0;
}
int snd_seq_fm_init(struct snd_seq_kinstr_ops *ops,
struct snd_seq_kinstr_ops *next)
{
memset(ops, 0, sizeof(*ops));
// ops->private_data = private_data;
ops->add_len = sizeof(struct fm_instrument);
ops->instr_type = SNDRV_SEQ_INSTR_ID_OPL2_3;
ops->put = snd_seq_fm_put;
ops->get = snd_seq_fm_get;
ops->get_size = snd_seq_fm_get_size;
// ops->remove = snd_seq_fm_remove;
// ops->notify = snd_seq_fm_notify;
ops->next = next;
return 0;
}
/*
* Init part
*/
static int __init alsa_ainstr_fm_init(void)
{
return 0;
}
static void __exit alsa_ainstr_fm_exit(void)
{
}
module_init(alsa_ainstr_fm_init)
module_exit(alsa_ainstr_fm_exit)
EXPORT_SYMBOL(snd_seq_fm_init);

359
sound/core/seq/instr/ainstr_gf1.c
View File

@ -1,359 +0,0 @@
/*
* GF1 (GUS) Patch - Instrument routines
* 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
* 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/init.h>
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/ainstr_gf1.h>
#include <sound/initval.h>
#include <asm/uaccess.h>
MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Advanced Linux Sound Architecture GF1 (GUS) Patch support.");
MODULE_LICENSE("GPL");
static unsigned int snd_seq_gf1_size(unsigned int size, unsigned int format)
{
unsigned int result = size;
if (format & GF1_WAVE_16BIT)
result <<= 1;
if (format & GF1_WAVE_STEREO)
result <<= 1;
return format;
}
static int snd_seq_gf1_copy_wave_from_stream(struct snd_gf1_ops *ops,
struct gf1_instrument *ip,
char __user **data,
long *len,
int atomic)
{
struct gf1_wave *wp, *prev;
struct gf1_xwave xp;
int err;
gfp_t gfp_mask;
unsigned int real_size;
gfp_mask = atomic ? GFP_ATOMIC : GFP_KERNEL;
if (*len < (long)sizeof(xp))
return -EINVAL;
if (copy_from_user(&xp, *data, sizeof(xp)))
return -EFAULT;
*data += sizeof(xp);
*len -= sizeof(xp);
wp = kzalloc(sizeof(*wp), gfp_mask);
if (wp == NULL)
return -ENOMEM;
wp->share_id[0] = le32_to_cpu(xp.share_id[0]);
wp->share_id[1] = le32_to_cpu(xp.share_id[1]);
wp->share_id[2] = le32_to_cpu(xp.share_id[2]);
wp->share_id[3] = le32_to_cpu(xp.share_id[3]);
wp->format = le32_to_cpu(xp.format);
wp->size = le32_to_cpu(xp.size);
wp->start = le32_to_cpu(xp.start);
wp->loop_start = le32_to_cpu(xp.loop_start);
wp->loop_end = le32_to_cpu(xp.loop_end);
wp->loop_repeat = le16_to_cpu(xp.loop_repeat);
wp->flags = xp.flags;
wp->sample_rate = le32_to_cpu(xp.sample_rate);
wp->low_frequency = le32_to_cpu(xp.low_frequency);
wp->high_frequency = le32_to_cpu(xp.high_frequency);
wp->root_frequency = le32_to_cpu(xp.root_frequency);
wp->tune = le16_to_cpu(xp.tune);
wp->balance = xp.balance;
memcpy(wp->envelope_rate, xp.envelope_rate, 6);
memcpy(wp->envelope_offset, xp.envelope_offset, 6);
wp->tremolo_sweep = xp.tremolo_sweep;
wp->tremolo_rate = xp.tremolo_rate;
wp->tremolo_depth = xp.tremolo_depth;
wp->vibrato_sweep = xp.vibrato_sweep;
wp->vibrato_rate = xp.vibrato_rate;
wp->vibrato_depth = xp.vibrato_depth;
wp->scale_frequency = le16_to_cpu(xp.scale_frequency);
wp->scale_factor = le16_to_cpu(xp.scale_factor);
real_size = snd_seq_gf1_size(wp->size, wp->format);
if ((long)real_size > *len) {
kfree(wp);
return -ENOMEM;
}
if (ops->put_sample) {
err = ops->put_sample(ops->private_data, wp,
*data, real_size, atomic);
if (err < 0) {
kfree(wp);
return err;
}
}
*data += real_size;
*len -= real_size;
prev = ip->wave;
if (prev) {
while (prev->next) prev = prev->next;
prev->next = wp;
} else {
ip->wave = wp;
}
return 0;
}
static void snd_seq_gf1_wave_free(struct snd_gf1_ops *ops,
struct gf1_wave *wave,
int atomic)
{
if (ops->remove_sample)
ops->remove_sample(ops->private_data, wave, atomic);
kfree(wave);
}
static void snd_seq_gf1_instr_free(struct snd_gf1_ops *ops,
struct gf1_instrument *ip,
int atomic)
{
struct gf1_wave *wave;
while ((wave = ip->wave) != NULL) {
ip->wave = wave->next;
snd_seq_gf1_wave_free(ops, wave, atomic);
}
}
static int snd_seq_gf1_put(void *private_data, struct snd_seq_kinstr *instr,
char __user *instr_data, long len, int atomic,
int cmd)
{
struct snd_gf1_ops *ops = private_data;
struct gf1_instrument *ip;
struct gf1_xinstrument ix;
int err;
gfp_t gfp_mask;
if (cmd != SNDRV_SEQ_INSTR_PUT_CMD_CREATE)
return -EINVAL;
gfp_mask = atomic ? GFP_ATOMIC : GFP_KERNEL;
/* copy instrument data */
if (len < (long)sizeof(ix))
return -EINVAL;
if (copy_from_user(&ix, instr_data, sizeof(ix)))
return -EFAULT;
if (ix.stype != GF1_STRU_INSTR)
return -EINVAL;
instr_data += sizeof(ix);
len -= sizeof(ix);
ip = (struct gf1_instrument *)KINSTR_DATA(instr);
ip->exclusion = le16_to_cpu(ix.exclusion);
ip->exclusion_group = le16_to_cpu(ix.exclusion_group);
ip->effect1 = ix.effect1;
ip->effect1_depth = ix.effect1_depth;
ip->effect2 = ix.effect2;
ip->effect2_depth = ix.effect2_depth;
/* copy layers */
while (len > (long)sizeof(__u32)) {
__u32 stype;
if (copy_from_user(&stype, instr_data, sizeof(stype)))
return -EFAULT;
if (stype != GF1_STRU_WAVE) {
snd_seq_gf1_instr_free(ops, ip, atomic);
return -EINVAL;
}
err = snd_seq_gf1_copy_wave_from_stream(ops,
ip,
&instr_data,
&len,
atomic);
if (err < 0) {
snd_seq_gf1_instr_free(ops, ip, atomic);
return err;
}
}
return 0;
}
static int snd_seq_gf1_copy_wave_to_stream(struct snd_gf1_ops *ops,
struct gf1_instrument *ip,
char __user **data,
long *len,
int atomic)
{
struct gf1_wave *wp;
struct gf1_xwave xp;
int err;
unsigned int real_size;
for (wp = ip->wave; wp; wp = wp->next) {
if (*len < (long)sizeof(xp))
return -ENOMEM;
memset(&xp, 0, sizeof(xp));
xp.stype = GF1_STRU_WAVE;
xp.share_id[0] = cpu_to_le32(wp->share_id[0]);
xp.share_id[1] = cpu_to_le32(wp->share_id[1]);
xp.share_id[2] = cpu_to_le32(wp->share_id[2]);
xp.share_id[3] = cpu_to_le32(wp->share_id[3]);
xp.format = cpu_to_le32(wp->format);
xp.size = cpu_to_le32(wp->size);
xp.start = cpu_to_le32(wp->start);
xp.loop_start = cpu_to_le32(wp->loop_start);
xp.loop_end = cpu_to_le32(wp->loop_end);
xp.loop_repeat = cpu_to_le32(wp->loop_repeat);
xp.flags = wp->flags;
xp.sample_rate = cpu_to_le32(wp->sample_rate);
xp.low_frequency = cpu_to_le32(wp->low_frequency);
xp.high_frequency = cpu_to_le32(wp->high_frequency);
xp.root_frequency = cpu_to_le32(wp->root_frequency);
xp.tune = cpu_to_le16(wp->tune);
xp.balance = wp->balance;
memcpy(xp.envelope_rate, wp->envelope_rate, 6);
memcpy(xp.envelope_offset, wp->envelope_offset, 6);
xp.tremolo_sweep = wp->tremolo_sweep;
xp.tremolo_rate = wp->tremolo_rate;
xp.tremolo_depth = wp->tremolo_depth;
xp.vibrato_sweep = wp->vibrato_sweep;
xp.vibrato_rate = wp->vibrato_rate;
xp.vibrato_depth = wp->vibrato_depth;
xp.scale_frequency = cpu_to_le16(wp->scale_frequency);
xp.scale_factor = cpu_to_le16(wp->scale_factor);
if (copy_to_user(*data, &xp, sizeof(xp)))
return -EFAULT;
*data += sizeof(xp);
*len -= sizeof(xp);
real_size = snd_seq_gf1_size(wp->size, wp->format);
if (*len < (long)real_size)
return -ENOMEM;
if (ops->get_sample) {
err = ops->get_sample(ops->private_data, wp,
*data, real_size, atomic);
if (err < 0)
return err;
}
*data += wp->size;
*len -= wp->size;
}
return 0;
}
static int snd_seq_gf1_get(void *private_data, struct snd_seq_kinstr *instr,
char __user *instr_data, long len, int atomic,
int cmd)
{
struct snd_gf1_ops *ops = private_data;
struct gf1_instrument *ip;
struct gf1_xinstrument ix;
if (cmd != SNDRV_SEQ_INSTR_GET_CMD_FULL)
return -EINVAL;
if (len < (long)sizeof(ix))
return -ENOMEM;
memset(&ix, 0, sizeof(ix));
ip = (struct gf1_instrument *)KINSTR_DATA(instr);
ix.stype = GF1_STRU_INSTR;
ix.exclusion = cpu_to_le16(ip->exclusion);
ix.exclusion_group = cpu_to_le16(ip->exclusion_group);
ix.effect1 = cpu_to_le16(ip->effect1);
ix.effect1_depth = cpu_to_le16(ip->effect1_depth);
ix.effect2 = ip->effect2;
ix.effect2_depth = ip->effect2_depth;
if (copy_to_user(instr_data, &ix, sizeof(ix)))
return -EFAULT;
instr_data += sizeof(ix);
len -= sizeof(ix);
return snd_seq_gf1_copy_wave_to_stream(ops,
ip,
&instr_data,
&len,
atomic);
}
static int snd_seq_gf1_get_size(void *private_data, struct snd_seq_kinstr *instr,
long *size)
{
long result;
struct gf1_instrument *ip;
struct gf1_wave *wp;
*size = 0;
ip = (struct gf1_instrument *)KINSTR_DATA(instr);
result = sizeof(struct gf1_xinstrument);
for (wp = ip->wave; wp; wp = wp->next) {
result += sizeof(struct gf1_xwave);
result += wp->size;
}
*size = result;
return 0;
}
static int snd_seq_gf1_remove(void *private_data,
struct snd_seq_kinstr *instr,
int atomic)
{
struct snd_gf1_ops *ops = private_data;
struct gf1_instrument *ip;
ip = (struct gf1_instrument *)KINSTR_DATA(instr);
snd_seq_gf1_instr_free(ops, ip, atomic);
return 0;
}
static void snd_seq_gf1_notify(void *private_data,
struct snd_seq_kinstr *instr,
int what)
{
struct snd_gf1_ops *ops = private_data;
if (ops->notify)
ops->notify(ops->private_data, instr, what);
}
int snd_seq_gf1_init(struct snd_gf1_ops *ops,
void *private_data,
struct snd_seq_kinstr_ops *next)
{
memset(ops, 0, sizeof(*ops));
ops->private_data = private_data;
ops->kops.private_data = ops;
ops->kops.add_len = sizeof(struct gf1_instrument);
ops->kops.instr_type = SNDRV_SEQ_INSTR_ID_GUS_PATCH;
ops->kops.put = snd_seq_gf1_put;
ops->kops.get = snd_seq_gf1_get;
ops->kops.get_size = snd_seq_gf1_get_size;
ops->kops.remove = snd_seq_gf1_remove;
ops->kops.notify = snd_seq_gf1_notify;
ops->kops.next = next;
return 0;
}
/*
* Init part
*/
static int __init alsa_ainstr_gf1_init(void)
{
return 0;
}
static void __exit alsa_ainstr_gf1_exit(void)
{
}
module_init(alsa_ainstr_gf1_init)
module_exit(alsa_ainstr_gf1_exit)
EXPORT_SYMBOL(snd_seq_gf1_init);

623
sound/core/seq/instr/ainstr_iw.c
View File

@ -1,623 +0,0 @@
/*
* IWFFFF - AMD InterWave (tm) - Instrument routines
* 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
* 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/init.h>
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/ainstr_iw.h>
#include <sound/initval.h>
#include <asm/uaccess.h>
MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Advanced Linux Sound Architecture IWFFFF support.");
MODULE_LICENSE("GPL");
static unsigned int snd_seq_iwffff_size(unsigned int size, unsigned int format)
{
unsigned int result = size;
if (format & IWFFFF_WAVE_16BIT)
result <<= 1;
if (format & IWFFFF_WAVE_STEREO)
result <<= 1;
return result;
}
static void snd_seq_iwffff_copy_lfo_from_stream(struct iwffff_lfo *fp,
struct iwffff_xlfo *fx)
{
fp->freq = le16_to_cpu(fx->freq);
fp->depth = le16_to_cpu(fx->depth);
fp->sweep = le16_to_cpu(fx->sweep);
fp->shape = fx->shape;
fp->delay = fx->delay;
}
static int snd_seq_iwffff_copy_env_from_stream(__u32 req_stype,
struct iwffff_layer *lp,
struct iwffff_env *ep,
struct iwffff_xenv *ex,
char __user **data,
long *len,
gfp_t gfp_mask)
{
__u32 stype;
struct iwffff_env_record *rp, *rp_last;
struct iwffff_xenv_record rx;
struct iwffff_env_point *pp;
struct iwffff_xenv_point px;
int points_size, idx;
ep->flags = ex->flags;
ep->mode = ex->mode;
ep->index = ex->index;
rp_last = NULL;
while (1) {
if (*len < (long)sizeof(__u32))
return -EINVAL;
if (copy_from_user(&stype, *data, sizeof(stype)))
return -EFAULT;
if (stype == IWFFFF_STRU_WAVE)
return 0;
if (req_stype != stype) {
if (stype == IWFFFF_STRU_ENV_RECP ||
stype == IWFFFF_STRU_ENV_RECV)
return 0;
}
if (*len < (long)sizeof(rx))
return -EINVAL;
if (copy_from_user(&rx, *data, sizeof(rx)))
return -EFAULT;
*data += sizeof(rx);
*len -= sizeof(rx);
points_size = (le16_to_cpu(rx.nattack) + le16_to_cpu(rx.nrelease)) * 2 * sizeof(__u16);
if (points_size > *len)
return -EINVAL;
rp = kzalloc(sizeof(*rp) + points_size, gfp_mask);
if (rp == NULL)
return -ENOMEM;
rp->nattack = le16_to_cpu(rx.nattack);
rp->nrelease = le16_to_cpu(rx.nrelease);
rp->sustain_offset = le16_to_cpu(rx.sustain_offset);
rp->sustain_rate = le16_to_cpu(rx.sustain_rate);
rp->release_rate = le16_to_cpu(rx.release_rate);
rp->hirange = rx.hirange;
pp = (struct iwffff_env_point *)(rp + 1);
for (idx = 0; idx < rp->nattack + rp->nrelease; idx++) {
if (copy_from_user(&px, *data, sizeof(px)))
return -EFAULT;
*data += sizeof(px);
*len -= sizeof(px);
pp->offset = le16_to_cpu(px.offset);
pp->rate = le16_to_cpu(px.rate);
}
if (ep->record == NULL) {
ep->record = rp;
} else {
rp_last = rp;
}
rp_last = rp;
}
return 0;
}
static int snd_seq_iwffff_copy_wave_from_stream(struct snd_iwffff_ops *ops,
struct iwffff_layer *lp,
char __user **data,
long *len,
int atomic)
{
struct iwffff_wave *wp, *prev;
struct iwffff_xwave xp;
int err;
gfp_t gfp_mask;
unsigned int real_size;
gfp_mask = atomic ? GFP_ATOMIC : GFP_KERNEL;
if (*len < (long)sizeof(xp))
return -EINVAL;
if (copy_from_user(&xp, *data, sizeof(xp)))
return -EFAULT;
*data += sizeof(xp);
*len -= sizeof(xp);
wp = kzalloc(sizeof(*wp), gfp_mask);
if (wp == NULL)
return -ENOMEM;
wp->share_id[0] = le32_to_cpu(xp.share_id[0]);
wp->share_id[1] = le32_to_cpu(xp.share_id[1]);
wp->share_id[2] = le32_to_cpu(xp.share_id[2]);
wp->share_id[3] = le32_to_cpu(xp.share_id[3]);
wp->format = le32_to_cpu(xp.format);
wp->address.memory = le32_to_cpu(xp.offset);
wp->size = le32_to_cpu(xp.size);
wp->start = le32_to_cpu(xp.start);
wp->loop_start = le32_to_cpu(xp.loop_start);
wp->loop_end = le32_to_cpu(xp.loop_end);
wp->loop_repeat = le16_to_cpu(xp.loop_repeat);
wp->sample_ratio = le32_to_cpu(xp.sample_ratio);
wp->attenuation = xp.attenuation;
wp->low_note = xp.low_note;
wp->high_note = xp.high_note;
real_size = snd_seq_iwffff_size(wp->size, wp->format);
if (!(wp->format & IWFFFF_WAVE_ROM)) {
if ((long)real_size > *len) {
kfree(wp);
return -ENOMEM;
}
}
if (ops->put_sample) {
err = ops->put_sample(ops->private_data, wp,
*data, real_size, atomic);
if (err < 0) {
kfree(wp);
return err;
}
}
if (!(wp->format & IWFFFF_WAVE_ROM)) {
*data += real_size;
*len -= real_size;
}
prev = lp->wave;
if (prev) {
while (prev->next) prev = prev->next;
prev->next = wp;
} else {
lp->wave = wp;
}
return 0;
}
static void snd_seq_iwffff_env_free(struct snd_iwffff_ops *ops,
struct iwffff_env *env,
int atomic)
{
struct iwffff_env_record *rec;
while ((rec = env->record) != NULL) {
env->record = rec->next;
kfree(rec);
}
}
static void snd_seq_iwffff_wave_free(struct snd_iwffff_ops *ops,
struct iwffff_wave *wave,
int atomic)
{
if (ops->remove_sample)
ops->remove_sample(ops->private_data, wave, atomic);
kfree(wave);
}
static void snd_seq_iwffff_instr_free(struct snd_iwffff_ops *ops,
struct iwffff_instrument *ip,
int atomic)
{
struct iwffff_layer *layer;
struct iwffff_wave *wave;
while ((layer = ip->layer) != NULL) {
ip->layer = layer->next;
snd_seq_iwffff_env_free(ops, &layer->penv, atomic);
snd_seq_iwffff_env_free(ops, &layer->venv, atomic);
while ((wave = layer->wave) != NULL) {
layer->wave = wave->next;
snd_seq_iwffff_wave_free(ops, wave, atomic);
}
kfree(layer);
}
}
static int snd_seq_iwffff_put(void *private_data, struct snd_seq_kinstr *instr,
char __user *instr_data, long len, int atomic,
int cmd)
{
struct snd_iwffff_ops *ops = private_data;
struct iwffff_instrument *ip;
struct iwffff_xinstrument ix;
struct iwffff_layer *lp, *prev_lp;
struct iwffff_xlayer lx;
int err;
gfp_t gfp_mask;
if (cmd != SNDRV_SEQ_INSTR_PUT_CMD_CREATE)
return -EINVAL;
gfp_mask = atomic ? GFP_ATOMIC : GFP_KERNEL;
/* copy instrument data */
if (len < (long)sizeof(ix))
return -EINVAL;
if (copy_from_user(&ix, instr_data, sizeof(ix)))
return -EFAULT;
if (ix.stype != IWFFFF_STRU_INSTR)
return -EINVAL;
instr_data += sizeof(ix);
len -= sizeof(ix);
ip = (struct iwffff_instrument *)KINSTR_DATA(instr);
ip->exclusion = le16_to_cpu(ix.exclusion);
ip->layer_type = le16_to_cpu(ix.layer_type);
ip->exclusion_group = le16_to_cpu(ix.exclusion_group);
ip->effect1 = ix.effect1;
ip->effect1_depth = ix.effect1_depth;
ip->effect2 = ix.effect2;
ip->effect2_depth = ix.effect2_depth;
/* copy layers */
prev_lp = NULL;
while (len > 0) {
if (len < (long)sizeof(struct iwffff_xlayer)) {
snd_seq_iwffff_instr_free(ops, ip, atomic);
return -EINVAL;
}
if (copy_from_user(&lx, instr_data, sizeof(lx)))
return -EFAULT;
instr_data += sizeof(lx);
len -= sizeof(lx);
if (lx.stype != IWFFFF_STRU_LAYER) {
snd_seq_iwffff_instr_free(ops, ip, atomic);
return -EINVAL;
}
lp = kzalloc(sizeof(*lp), gfp_mask);
if (lp == NULL) {
snd_seq_iwffff_instr_free(ops, ip, atomic);
return -ENOMEM;
}
if (prev_lp) {
prev_lp->next = lp;
} else {
ip->layer = lp;
}
prev_lp = lp;
lp->flags = lx.flags;
lp->velocity_mode = lx.velocity_mode;
lp->layer_event = lx.layer_event;
lp->low_range = lx.low_range;
lp->high_range = lx.high_range;
lp->pan = lx.pan;
lp->pan_freq_scale = lx.pan_freq_scale;
lp->attenuation = lx.attenuation;
snd_seq_iwffff_copy_lfo_from_stream(&lp->tremolo, &lx.tremolo);
snd_seq_iwffff_copy_lfo_from_stream(&lp->vibrato, &lx.vibrato);
lp->freq_scale = le16_to_cpu(lx.freq_scale);
lp->freq_center = lx.freq_center;
err = snd_seq_iwffff_copy_env_from_stream(IWFFFF_STRU_ENV_RECP,
lp,
&lp->penv, &lx.penv,
&instr_data, &len,
gfp_mask);
if (err < 0) {
snd_seq_iwffff_instr_free(ops, ip, atomic);
return err;
}
err = snd_seq_iwffff_copy_env_from_stream(IWFFFF_STRU_ENV_RECV,
lp,
&lp->venv, &lx.venv,
&instr_data, &len,
gfp_mask);
if (err < 0) {
snd_seq_iwffff_instr_free(ops, ip, atomic);
return err;
}
while (len > (long)sizeof(__u32)) {
__u32 stype;
if (copy_from_user(&stype, instr_data, sizeof(stype)))
return -EFAULT;
if (stype != IWFFFF_STRU_WAVE)
break;
err = snd_seq_iwffff_copy_wave_from_stream(ops,
lp,
&instr_data,
&len,
atomic);
if (err < 0) {
snd_seq_iwffff_instr_free(ops, ip, atomic);
return err;
}
}
}
return 0;
}
static void snd_seq_iwffff_copy_lfo_to_stream(struct iwffff_xlfo *fx,
struct iwffff_lfo *fp)
{
fx->freq = cpu_to_le16(fp->freq);
fx->depth = cpu_to_le16(fp->depth);
fx->sweep = cpu_to_le16(fp->sweep);
fp->shape = fx->shape;
fp->delay = fx->delay;
}
static int snd_seq_iwffff_copy_env_to_stream(__u32 req_stype,
struct iwffff_layer *lp,
struct iwffff_xenv *ex,
struct iwffff_env *ep,
char __user **data,
long *len)
{
struct iwffff_env_record *rp;
struct iwffff_xenv_record rx;
struct iwffff_env_point *pp;
struct iwffff_xenv_point px;
int points_size, idx;
ex->flags = ep->flags;
ex->mode = ep->mode;
ex->index = ep->index;
for (rp = ep->record; rp; rp = rp->next) {
if (*len < (long)sizeof(rx))
return -ENOMEM;
memset(&rx, 0, sizeof(rx));
rx.stype = req_stype;
rx.nattack = cpu_to_le16(rp->nattack);
rx.nrelease = cpu_to_le16(rp->nrelease);
rx.sustain_offset = cpu_to_le16(rp->sustain_offset);
rx.sustain_rate = cpu_to_le16(rp->sustain_rate);
rx.release_rate = cpu_to_le16(rp->release_rate);
rx.hirange = cpu_to_le16(rp->hirange);
if (copy_to_user(*data, &rx, sizeof(rx)))
return -EFAULT;
*data += sizeof(rx);
*len -= sizeof(rx);
points_size = (rp->nattack + rp->nrelease) * 2 * sizeof(__u16);
if (*len < points_size)
return -ENOMEM;
pp = (struct iwffff_env_point *)(rp + 1);
for (idx = 0; idx < rp->nattack + rp->nrelease; idx++) {
px.offset = cpu_to_le16(pp->offset);
px.rate = cpu_to_le16(pp->rate);
if (copy_to_user(*data, &px, sizeof(px)))
return -EFAULT;
*data += sizeof(px);
*len -= sizeof(px);
}
}
return 0;
}
static int snd_seq_iwffff_copy_wave_to_stream(struct snd_iwffff_ops *ops,
struct iwffff_layer *lp,
char __user **data,
long *len,
int atomic)
{
struct iwffff_wave *wp;
struct iwffff_xwave xp;
int err;
unsigned int real_size;
for (wp = lp->wave; wp; wp = wp->next) {
if (*len < (long)sizeof(xp))
return -ENOMEM;
memset(&xp, 0, sizeof(xp));
xp.stype = IWFFFF_STRU_WAVE;
xp.share_id[0] = cpu_to_le32(wp->share_id[0]);
xp.share_id[1] = cpu_to_le32(wp->share_id[1]);
xp.share_id[2] = cpu_to_le32(wp->share_id[2]);
xp.share_id[3] = cpu_to_le32(wp->share_id[3]);
xp.format = cpu_to_le32(wp->format);
if (wp->format & IWFFFF_WAVE_ROM)
xp.offset = cpu_to_le32(wp->address.memory);
xp.size = cpu_to_le32(wp->size);
xp.start = cpu_to_le32(wp->start);
xp.loop_start = cpu_to_le32(wp->loop_start);
xp.loop_end = cpu_to_le32(wp->loop_end);
xp.loop_repeat = cpu_to_le32(wp->loop_repeat);
xp.sample_ratio = cpu_to_le32(wp->sample_ratio);
xp.attenuation = wp->attenuation;
xp.low_note = wp->low_note;
xp.high_note = wp->high_note;
if (copy_to_user(*data, &xp, sizeof(xp)))
return -EFAULT;
*data += sizeof(xp);
*len -= sizeof(xp);
real_size = snd_seq_iwffff_size(wp->size, wp->format);
if (!(wp->format & IWFFFF_WAVE_ROM)) {
if (*len < (long)real_size)
return -ENOMEM;
}
if (ops->get_sample) {
err = ops->get_sample(ops->private_data, wp,
*data, real_size, atomic);
if (err < 0)
return err;
}
if (!(wp->format & IWFFFF_WAVE_ROM)) {
*data += real_size;
*len -= real_size;
}
}
return 0;
}
static int snd_seq_iwffff_get(void *private_data, struct snd_seq_kinstr *instr,
char __user *instr_data, long len, int atomic, int cmd)
{
struct snd_iwffff_ops *ops = private_data;
struct iwffff_instrument *ip;
struct iwffff_xinstrument ix;
struct iwffff_layer *lp;
struct iwffff_xlayer lx;
char __user *layer_instr_data;
int err;
if (cmd != SNDRV_SEQ_INSTR_GET_CMD_FULL)
return -EINVAL;
if (len < (long)sizeof(ix))
return -ENOMEM;
memset(&ix, 0, sizeof(ix));
ip = (struct iwffff_instrument *)KINSTR_DATA(instr);
ix.stype = IWFFFF_STRU_INSTR;
ix.exclusion = cpu_to_le16(ip->exclusion);
ix.layer_type = cpu_to_le16(ip->layer_type);
ix.exclusion_group = cpu_to_le16(ip->exclusion_group);
ix.effect1 = cpu_to_le16(ip->effect1);
ix.effect1_depth = cpu_to_le16(ip->effect1_depth);
ix.effect2 = ip->effect2;
ix.effect2_depth = ip->effect2_depth;
if (copy_to_user(instr_data, &ix, sizeof(ix)))
return -EFAULT;
instr_data += sizeof(ix);
len -= sizeof(ix);
for (lp = ip->layer; lp; lp = lp->next) {
if (len < (long)sizeof(lx))
return -ENOMEM;
memset(&lx, 0, sizeof(lx));
lx.stype = IWFFFF_STRU_LAYER;
lx.flags = lp->flags;
lx.velocity_mode = lp->velocity_mode;
lx.layer_event = lp->layer_event;
lx.low_range = lp->low_range;
lx.high_range = lp->high_range;
lx.pan = lp->pan;
lx.pan_freq_scale = lp->pan_freq_scale;
lx.attenuation = lp->attenuation;
snd_seq_iwffff_copy_lfo_to_stream(&lx.tremolo, &lp->tremolo);
snd_seq_iwffff_copy_lfo_to_stream(&lx.vibrato, &lp->vibrato);
layer_instr_data = instr_data;
instr_data += sizeof(lx);
len -= sizeof(lx);
err = snd_seq_iwffff_copy_env_to_stream(IWFFFF_STRU_ENV_RECP,
lp,
&lx.penv, &lp->penv,
&instr_data, &len);
if (err < 0)
return err;
err = snd_seq_iwffff_copy_env_to_stream(IWFFFF_STRU_ENV_RECV,
lp,
&lx.venv, &lp->venv,
&instr_data, &len);
if (err < 0)
return err;
/* layer structure updating is now finished */
if (copy_to_user(layer_instr_data, &lx, sizeof(lx)))
return -EFAULT;
err = snd_seq_iwffff_copy_wave_to_stream(ops,
lp,
&instr_data,
&len,
atomic);
if (err < 0)
return err;
}
return 0;
}
static long snd_seq_iwffff_env_size_in_stream(struct iwffff_env *ep)
{
long result = 0;
struct iwffff_env_record *rp;
for (rp = ep->record; rp; rp = rp->next) {
result += sizeof(struct iwffff_xenv_record);
result += (rp->nattack + rp->nrelease) * 2 * sizeof(__u16);
}
return 0;
}
static long snd_seq_iwffff_wave_size_in_stream(struct iwffff_layer *lp)
{
long result = 0;
struct iwffff_wave *wp;
for (wp = lp->wave; wp; wp = wp->next) {
result += sizeof(struct iwffff_xwave);
if (!(wp->format & IWFFFF_WAVE_ROM))
result += wp->size;
}
return result;
}
static int snd_seq_iwffff_get_size(void *private_data, struct snd_seq_kinstr *instr,
long *size)
{
long result;
struct iwffff_instrument *ip;
struct iwffff_layer *lp;
*size = 0;
ip = (struct iwffff_instrument *)KINSTR_DATA(instr);
result = sizeof(struct iwffff_xinstrument);
for (lp = ip->layer; lp; lp = lp->next) {
result += sizeof(struct iwffff_xlayer);
result += snd_seq_iwffff_env_size_in_stream(&lp->penv);
result += snd_seq_iwffff_env_size_in_stream(&lp->venv);
result += snd_seq_iwffff_wave_size_in_stream(lp);
}
*size = result;
return 0;
}
static int snd_seq_iwffff_remove(void *private_data,
struct snd_seq_kinstr *instr,
int atomic)
{
struct snd_iwffff_ops *ops = private_data;
struct iwffff_instrument *ip;
ip = (struct iwffff_instrument *)KINSTR_DATA(instr);
snd_seq_iwffff_instr_free(ops, ip, atomic);
return 0;
}
static void snd_seq_iwffff_notify(void *private_data,
struct snd_seq_kinstr *instr,
int what)
{
struct snd_iwffff_ops *ops = private_data;
if (ops->notify)
ops->notify(ops->private_data, instr, what);
}
int snd_seq_iwffff_init(struct snd_iwffff_ops *ops,
void *private_data,
struct snd_seq_kinstr_ops *next)
{
memset(ops, 0, sizeof(*ops));
ops->private_data = private_data;
ops->kops.private_data = ops;
ops->kops.add_len = sizeof(struct iwffff_instrument);
ops->kops.instr_type = SNDRV_SEQ_INSTR_ID_INTERWAVE;
ops->kops.put = snd_seq_iwffff_put;
ops->kops.get = snd_seq_iwffff_get;
ops->kops.get_size = snd_seq_iwffff_get_size;
ops->kops.remove = snd_seq_iwffff_remove;
ops->kops.notify = snd_seq_iwffff_notify;
ops->kops.next = next;
return 0;
}
/*
* Init part
*/
static int __init alsa_ainstr_iw_init(void)
{
return 0;
}
static void __exit alsa_ainstr_iw_exit(void)
{
}
module_init(alsa_ainstr_iw_init)
module_exit(alsa_ainstr_iw_exit)
EXPORT_SYMBOL(snd_seq_iwffff_init);

215
sound/core/seq/instr/ainstr_simple.c
View File

@ -1,215 +0,0 @@
/*
* Simple (MOD player) - Instrument routines
* 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
* 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/init.h>
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/ainstr_simple.h>
#include <sound/initval.h>
#include <asm/uaccess.h>
MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Advanced Linux Sound Architecture Simple Instrument support.");
MODULE_LICENSE("GPL");
static unsigned int snd_seq_simple_size(unsigned int size, unsigned int format)
{
unsigned int result = size;
if (format & SIMPLE_WAVE_16BIT)
result <<= 1;
if (format & SIMPLE_WAVE_STEREO)
result <<= 1;
return result;
}
static void snd_seq_simple_instr_free(struct snd_simple_ops *ops,
struct simple_instrument *ip,
int atomic)
{
if (ops->remove_sample)
ops->remove_sample(ops->private_data, ip, atomic);
}
static int snd_seq_simple_put(void *private_data, struct snd_seq_kinstr *instr,
char __user *instr_data, long len,
int atomic, int cmd)
{
struct snd_simple_ops *ops = private_data;
struct simple_instrument *ip;
struct simple_xinstrument ix;
int err;
gfp_t gfp_mask;
unsigned int real_size;
if (cmd != SNDRV_SEQ_INSTR_PUT_CMD_CREATE)
return -EINVAL;
gfp_mask = atomic ? GFP_ATOMIC : GFP_KERNEL;
/* copy instrument data */
if (len < (long)sizeof(ix))
return -EINVAL;
if (copy_from_user(&ix, instr_data, sizeof(ix)))
return -EFAULT;
if (ix.stype != SIMPLE_STRU_INSTR)
return -EINVAL;
instr_data += sizeof(ix);
len -= sizeof(ix);
ip = (struct simple_instrument *)KINSTR_DATA(instr);
ip->share_id[0] = le32_to_cpu(ix.share_id[0]);
ip->share_id[1] = le32_to_cpu(ix.share_id[1]);
ip->share_id[2] = le32_to_cpu(ix.share_id[2]);
ip->share_id[3] = le32_to_cpu(ix.share_id[3]);
ip->format = le32_to_cpu(ix.format);
ip->size = le32_to_cpu(ix.size);
ip->start = le32_to_cpu(ix.start);
ip->loop_start = le32_to_cpu(ix.loop_start);
ip->loop_end = le32_to_cpu(ix.loop_end);
ip->loop_repeat = le16_to_cpu(ix.loop_repeat);
ip->effect1 = ix.effect1;
ip->effect1_depth = ix.effect1_depth;
ip->effect2 = ix.effect2;
ip->effect2_depth = ix.effect2_depth;
real_size = snd_seq_simple_size(ip->size, ip->format);
if (len < (long)real_size)
return -EINVAL;
if (ops->put_sample) {
err = ops->put_sample(ops->private_data, ip,
instr_data, real_size, atomic);
if (err < 0)
return err;
}
return 0;
}
static int snd_seq_simple_get(void *private_data, struct snd_seq_kinstr *instr,
char __user *instr_data, long len,
int atomic, int cmd)
{
struct snd_simple_ops *ops = private_data;
struct simple_instrument *ip;
struct simple_xinstrument ix;
int err;
unsigned int real_size;
if (cmd != SNDRV_SEQ_INSTR_GET_CMD_FULL)
return -EINVAL;
if (len < (long)sizeof(ix))
return -ENOMEM;
memset(&ix, 0, sizeof(ix));
ip = (struct simple_instrument *)KINSTR_DATA(instr);
ix.stype = SIMPLE_STRU_INSTR;
ix.share_id[0] = cpu_to_le32(ip->share_id[0]);
ix.share_id[1] = cpu_to_le32(ip->share_id[1]);
ix.share_id[2] = cpu_to_le32(ip->share_id[2]);
ix.share_id[3] = cpu_to_le32(ip->share_id[3]);
ix.format = cpu_to_le32(ip->format);
ix.size = cpu_to_le32(ip->size);
ix.start = cpu_to_le32(ip->start);
ix.loop_start = cpu_to_le32(ip->loop_start);
ix.loop_end = cpu_to_le32(ip->loop_end);
ix.loop_repeat = cpu_to_le32(ip->loop_repeat);
ix.effect1 = cpu_to_le16(ip->effect1);
ix.effect1_depth = cpu_to_le16(ip->effect1_depth);
ix.effect2 = ip->effect2;
ix.effect2_depth = ip->effect2_depth;
if (copy_to_user(instr_data, &ix, sizeof(ix)))
return -EFAULT;
instr_data += sizeof(ix);
len -= sizeof(ix);
real_size = snd_seq_simple_size(ip->size, ip->format);
if (len < (long)real_size)
return -ENOMEM;
if (ops->get_sample) {
err = ops->get_sample(ops->private_data, ip,
instr_data, real_size, atomic);
if (err < 0)
return err;
}
return 0;
}
static int snd_seq_simple_get_size(void *private_data, struct snd_seq_kinstr *instr,
long *size)
{
struct simple_instrument *ip;
ip = (struct simple_instrument *)KINSTR_DATA(instr);
*size = sizeof(struct simple_xinstrument) + snd_seq_simple_size(ip->size, ip->format);
return 0;
}
static int snd_seq_simple_remove(void *private_data,
struct snd_seq_kinstr *instr,
int atomic)
{
struct snd_simple_ops *ops = private_data;
struct simple_instrument *ip;
ip = (struct simple_instrument *)KINSTR_DATA(instr);
snd_seq_simple_instr_free(ops, ip, atomic);
return 0;
}
static void snd_seq_simple_notify(void *private_data,
struct snd_seq_kinstr *instr,
int what)
{
struct snd_simple_ops *ops = private_data;
if (ops->notify)
ops->notify(ops->private_data, instr, what);
}
int snd_seq_simple_init(struct snd_simple_ops *ops,
void *private_data,
struct snd_seq_kinstr_ops *next)
{
memset(ops, 0, sizeof(*ops));
ops->private_data = private_data;
ops->kops.private_data = ops;
ops->kops.add_len = sizeof(struct simple_instrument);
ops->kops.instr_type = SNDRV_SEQ_INSTR_ID_SIMPLE;
ops->kops.put = snd_seq_simple_put;
ops->kops.get = snd_seq_simple_get;
ops->kops.get_size = snd_seq_simple_get_size;
ops->kops.remove = snd_seq_simple_remove;
ops->kops.notify = snd_seq_simple_notify;
ops->kops.next = next;
return 0;
}
/*
* Init part
*/
static int __init alsa_ainstr_simple_init(void)
{
return 0;
}
static void __exit alsa_ainstr_simple_exit(void)
{
}
module_init(alsa_ainstr_simple_init)
module_exit(alsa_ainstr_simple_exit)
EXPORT_SYMBOL(snd_seq_simple_init);

1
sound/core/seq/oss/seq_oss.c
View File

@ -20,7 +20,6 @@
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include <sound/driver.h>
#include <linux/init.h>
#include <linux/moduleparam.h>
#include <linux/mutex.h>

1
sound/core/seq/oss/seq_oss_device.h
View File

@ -21,7 +21,6 @@
#ifndef __SEQ_OSS_DEVICE_H
#define __SEQ_OSS_DEVICE_H
#include <sound/driver.h>
#include <linux/time.h>
#include <linux/wait.h>
#include <linux/slab.h>

1
sound/core/seq/seq.c
View File

@ -19,7 +19,6 @@
*
*/
#include <sound/driver.h>
#include <linux/init.h>
#include <linux/moduleparam.h>
#include <sound/core.h>

6
sound/core/seq/seq_clientmgr.c
View File

@ -21,7 +21,6 @@
*
*/
#include <sound/driver.h>
#include <linux/init.h>
#include <linux/slab.h>
#include <sound/core.h>
@ -130,8 +129,6 @@ static struct snd_seq_client *clientptr(int clientid)
return clienttab[clientid];
}
extern int seq_client_load[];
struct snd_seq_client *snd_seq_client_use_ptr(int clientid)
{
unsigned long flags;
@ -966,8 +963,7 @@ static int check_event_type_and_length(struct snd_seq_event *ev)
return -EINVAL;
break;
case SNDRV_SEQ_EVENT_LENGTH_VARUSR:
if (! snd_seq_ev_is_instr_type(ev) ||
! snd_seq_ev_is_direct(ev))
if (! snd_seq_ev_is_direct(ev))
return -EINVAL;
break;
}

2
sound/core/seq/seq_clientmgr.h
View File

@ -98,4 +98,6 @@ int snd_seq_kernel_client_write_poll(int clientid, struct file *file, poll_table
int snd_seq_client_notify_subscription(int client, int port,
struct snd_seq_port_subscribe *info, int evtype);
extern int seq_client_load[15];
#endif

1
sound/core/seq/seq_device.c
View File

@ -36,7 +36,6 @@
*
*/
#include <sound/driver.h>
#include <linux/init.h>
#include <sound/core.h>
#include <sound/info.h>

1
sound/core/seq/seq_dummy.c
View File

@ -18,7 +18,6 @@
*
*/
#include <sound/driver.h>
#include <linux/init.h>
#include <linux/slab.h>
#include <linux/moduleparam.h>

1
sound/core/seq/seq_fifo.c
View File

@ -19,7 +19,6 @@
*
*/
#include <sound/driver.h>
#include <sound/core.h>
#include <linux/slab.h>
#include "seq_fifo.h"

1
sound/core/seq/seq_info.c
View File

@ -19,7 +19,6 @@
*
*/
#include <sound/driver.h>
#include <linux/init.h>
#include <sound/core.h>

655
sound/core/seq/seq_instr.c
View File

@ -1,655 +0,0 @@
/*
* Generic Instrument routines for ALSA sequencer
* 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
* 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/init.h>
#include <linux/slab.h>
#include <sound/core.h>
#include "seq_clientmgr.h"
#include <sound/seq_instr.h>
#include <sound/initval.h>
MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Advanced Linux Sound Architecture sequencer instrument library.");
MODULE_LICENSE("GPL");
static void snd_instr_lock_ops(struct snd_seq_kinstr_list *list)
{
if (!(list->flags & SNDRV_SEQ_INSTR_FLG_DIRECT)) {
spin_lock_irqsave(&list->ops_lock, list->ops_flags);
} else {
mutex_lock(&list->ops_mutex);
}
}
static void snd_instr_unlock_ops(struct snd_seq_kinstr_list *list)
{
if (!(list->flags & SNDRV_SEQ_INSTR_FLG_DIRECT)) {
spin_unlock_irqrestore(&list->ops_lock, list->ops_flags);
} else {
mutex_unlock(&list->ops_mutex);
}
}
static struct snd_seq_kinstr *snd_seq_instr_new(int add_len, int atomic)
{
struct snd_seq_kinstr *instr;
instr = kzalloc(sizeof(struct snd_seq_kinstr) + add_len, atomic ? GFP_ATOMIC : GFP_KERNEL);
if (instr == NULL)
return NULL;
instr->add_len = add_len;
return instr;
}
static int snd_seq_instr_free(struct snd_seq_kinstr *instr, int atomic)
{
int result = 0;
if (instr == NULL)
return -EINVAL;
if (instr->ops && instr->ops->remove)
result = instr->ops->remove(instr->ops->private_data, instr, 1);
if (!result)
kfree(instr);
return result;
}
struct snd_seq_kinstr_list *snd_seq_instr_list_new(void)
{
struct snd_seq_kinstr_list *list;
list = kzalloc(sizeof(struct snd_seq_kinstr_list), GFP_KERNEL);
if (list == NULL)
return NULL;
spin_lock_init(&list->lock);
spin_lock_init(&list->ops_lock);
mutex_init(&list->ops_mutex);
list->owner = -1;
return list;
}
void snd_seq_instr_list_free(struct snd_seq_kinstr_list **list_ptr)
{
struct snd_seq_kinstr_list *list;
struct snd_seq_kinstr *instr;
struct snd_seq_kcluster *cluster;
int idx;
unsigned long flags;
if (list_ptr == NULL)
return;
list = *list_ptr;
*list_ptr = NULL;
if (list == NULL)
return;
for (idx = 0; idx < SNDRV_SEQ_INSTR_HASH_SIZE; idx++) {
while ((instr = list->hash[idx]) != NULL) {
list->hash[idx] = instr->next;
list->count--;
spin_lock_irqsave(&list->lock, flags);
while (instr->use) {
spin_unlock_irqrestore(&list->lock, flags);
schedule_timeout_uninterruptible(1);
spin_lock_irqsave(&list->lock, flags);
}
spin_unlock_irqrestore(&list->lock, flags);
if (snd_seq_instr_free(instr, 0)<0)
snd_printk(KERN_WARNING "instrument free problem\n");
}
while ((cluster = list->chash[idx]) != NULL) {
list->chash[idx] = cluster->next;
list->ccount--;
kfree(cluster);
}
}
kfree(list);
}
static int instr_free_compare(struct snd_seq_kinstr *instr,
struct snd_seq_instr_header *ifree,
unsigned int client)
{
switch (ifree->cmd) {
case SNDRV_SEQ_INSTR_FREE_CMD_ALL:
/* all, except private for other clients */
if ((instr->instr.std & 0xff000000) == 0)
return 0;
if (((instr->instr.std >> 24) & 0xff) == client)
return 0;
return 1;
case SNDRV_SEQ_INSTR_FREE_CMD_PRIVATE:
/* all my private instruments */
if ((instr->instr.std & 0xff000000) == 0)
return 1;
if (((instr->instr.std >> 24) & 0xff) == client)
return 0;
return 1;
case SNDRV_SEQ_INSTR_FREE_CMD_CLUSTER:
/* all my private instruments */
if ((instr->instr.std & 0xff000000) == 0) {
if (instr->instr.cluster == ifree->id.cluster)
return 0;
return 1;
}
if (((instr->instr.std >> 24) & 0xff) == client) {
if (instr->instr.cluster == ifree->id.cluster)
return 0;
}
return 1;
}
return 1;
}
int snd_seq_instr_list_free_cond(struct snd_seq_kinstr_list *list,
struct snd_seq_instr_header *ifree,
int client,
int atomic)
{
struct snd_seq_kinstr *instr, *prev, *next, *flist;
int idx;
unsigned long flags;
snd_instr_lock_ops(list);
for (idx = 0; idx < SNDRV_SEQ_INSTR_HASH_SIZE; idx++) {
spin_lock_irqsave(&list->lock, flags);
instr = list->hash[idx];
prev = flist = NULL;
while (instr) {
while (instr && instr_free_compare(instr, ifree, (unsigned int)client)) {
prev = instr;
instr = instr->next;
}
if (instr == NULL)
continue;
if (instr->ops && instr->ops->notify)
instr->ops->notify(instr->ops->private_data, instr, SNDRV_SEQ_INSTR_NOTIFY_REMOVE);
next = instr->next;
if (prev == NULL) {
list->hash[idx] = next;
} else {
prev->next = next;
}
list->count--;
instr->next = flist;
flist = instr;
instr = next;
}
spin_unlock_irqrestore(&list->lock, flags);
while (flist) {
instr = flist;
flist = instr->next;
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;
}
}
snd_instr_unlock_ops(list);
return 0;
}
static int compute_hash_instr_key(struct snd_seq_instr *instr)
{
int result;
result = instr->bank | (instr->prg << 16);
result += result >> 24;
result += result >> 16;
result += result >> 8;
return result & (SNDRV_SEQ_INSTR_HASH_SIZE-1);
}
#if 0
static int compute_hash_cluster_key(snd_seq_instr_cluster_t cluster)
{
int result;
result = cluster;
result += result >> 24;
result += result >> 16;
result += result >> 8;
return result & (SNDRV_SEQ_INSTR_HASH_SIZE-1);
}
#endif
static int compare_instr(struct snd_seq_instr *i1, struct snd_seq_instr *i2, int exact)
{
if (exact) {
if (i1->cluster != i2->cluster ||
i1->bank != i2->bank ||
i1->prg != i2->prg)
return 1;
if ((i1->std & 0xff000000) != (i2->std & 0xff000000))
return 1;
if (!(i1->std & i2->std))
return 1;
return 0;
} else {
unsigned int client_check;
if (i2->cluster && i1->cluster != i2->cluster)
return 1;
client_check = i2->std & 0xff000000;
if (client_check) {
if ((i1->std & 0xff000000) != client_check)
return 1;
} else {
if ((i1->std & i2->std) != i2->std)
return 1;
}
return i1->bank != i2->bank || i1->prg != i2->prg;
}
}
struct snd_seq_kinstr *snd_seq_instr_find(struct snd_seq_kinstr_list *list,
struct snd_seq_instr *instr,
int exact,
int follow_alias)
{
unsigned long flags;
int depth = 0;
struct snd_seq_kinstr *result;
if (list == NULL || instr == NULL)
return NULL;
spin_lock_irqsave(&list->lock, flags);
__again:
result = list->hash[compute_hash_instr_key(instr)];
while (result) {
if (!compare_instr(&result->instr, instr, exact)) {
if (follow_alias && (result->type == SNDRV_SEQ_INSTR_ATYPE_ALIAS)) {
instr = (struct snd_seq_instr *)KINSTR_DATA(result);
if (++depth > 10)
goto __not_found;
goto __again;
}
result->use++;
spin_unlock_irqrestore(&list->lock, flags);
return result;
}
result = result->next;
}
__not_found:
spin_unlock_irqrestore(&list->lock, flags);
return NULL;
}
void snd_seq_instr_free_use(struct snd_seq_kinstr_list *list,
struct snd_seq_kinstr *instr)
{
unsigned long flags;
if (list == NULL || instr == NULL)
return;
spin_lock_irqsave(&list->lock, flags);
if (instr->use <= 0) {
snd_printk(KERN_ERR "free_use: fatal!!! use = %i, name = '%s'\n", instr->use, instr->name);
} else {
instr->use--;
}
spin_unlock_irqrestore(&list->lock, flags);
}
static struct snd_seq_kinstr_ops *instr_ops(struct snd_seq_kinstr_ops *ops,
char *instr_type)
{
while (ops) {
if (!strcmp(ops->instr_type, instr_type))
return ops;
ops = ops->next;
}
return NULL;
}
static int instr_result(struct snd_seq_event *ev,
int type, int result,
int atomic)
{
struct snd_seq_event sev;
memset(&sev, 0, sizeof(sev));
sev.type = SNDRV_SEQ_EVENT_RESULT;
sev.flags = SNDRV_SEQ_TIME_STAMP_REAL | SNDRV_SEQ_EVENT_LENGTH_FIXED |
SNDRV_SEQ_PRIORITY_NORMAL;
sev.source = ev->dest;
sev.dest = ev->source;
sev.data.result.event = type;
sev.data.result.result = result;
#if 0
printk("instr result - type = %i, result = %i, queue = %i, source.client:port = %i:%i, dest.client:port = %i:%i\n",
type, result,
sev.queue,
sev.source.client, sev.source.port,
sev.dest.client, sev.dest.port);
#endif
return snd_seq_kernel_client_dispatch(sev.source.client, &sev, atomic, 0);
}
static int instr_begin(struct snd_seq_kinstr_ops *ops,
struct snd_seq_kinstr_list *list,
struct snd_seq_event *ev,
int atomic, int hop)
{
unsigned long flags;
spin_lock_irqsave(&list->lock, flags);
if (list->owner >= 0 && list->owner != ev->source.client) {
spin_unlock_irqrestore(&list->lock, flags);
return instr_result(ev, SNDRV_SEQ_EVENT_INSTR_BEGIN, -EBUSY, atomic);
}
list->owner = ev->source.client;
spin_unlock_irqrestore(&list->lock, flags);
return instr_result(ev, SNDRV_SEQ_EVENT_INSTR_BEGIN, 0, atomic);
}
static int instr_end(struct snd_seq_kinstr_ops *ops,
struct snd_seq_kinstr_list *list,
struct snd_seq_event *ev,
int atomic, int hop)
{
unsigned long flags;
/* TODO: timeout handling */
spin_lock_irqsave(&list->lock, flags);
if (list->owner == ev->source.client) {
list->owner = -1;
spin_unlock_irqrestore(&list->lock, flags);
return instr_result(ev, SNDRV_SEQ_EVENT_INSTR_END, 0, atomic);
}
spin_unlock_irqrestore(&list->lock, flags);
return instr_result(ev, SNDRV_SEQ_EVENT_INSTR_END, -EINVAL, atomic);
}
static int instr_info(struct snd_seq_kinstr_ops *ops,
struct snd_seq_kinstr_list *list,
struct snd_seq_event *ev,
int atomic, int hop)
{
return -ENXIO;
}
static int instr_format_info(struct snd_seq_kinstr_ops *ops,
struct snd_seq_kinstr_list *list,
struct snd_seq_event *ev,
int atomic, int hop)
{
return -ENXIO;
}
static int instr_reset(struct snd_seq_kinstr_ops *ops,
struct snd_seq_kinstr_list *list,
struct snd_seq_event *ev,
int atomic, int hop)
{
return -ENXIO;
}
static int instr_status(struct snd_seq_kinstr_ops *ops,
struct snd_seq_kinstr_list *list,
struct snd_seq_event *ev,
int atomic, int hop)
{
return -ENXIO;
}
static int instr_put(struct snd_seq_kinstr_ops *ops,
struct snd_seq_kinstr_list *list,
struct snd_seq_event *ev,
int atomic, int hop)
{
unsigned long flags;
struct snd_seq_instr_header put;
struct snd_seq_kinstr *instr;
int result = -EINVAL, len, key;
if ((ev->flags & SNDRV_SEQ_EVENT_LENGTH_MASK) != SNDRV_SEQ_EVENT_LENGTH_VARUSR)
goto __return;
if (ev->data.ext.len < sizeof(struct snd_seq_instr_header))
goto __return;
if (copy_from_user(&put, (void __user *)ev->data.ext.ptr,
sizeof(struct snd_seq_instr_header))) {
result = -EFAULT;
goto __return;
}
snd_instr_lock_ops(list);
if (put.id.instr.std & 0xff000000) { /* private instrument */
put.id.instr.std &= 0x00ffffff;
put.id.instr.std |= (unsigned int)ev->source.client << 24;
}
if ((instr = snd_seq_instr_find(list, &put.id.instr, 1, 0))) {
snd_seq_instr_free_use(list, instr);
snd_instr_unlock_ops(list);
result = -EBUSY;
goto __return;
}
ops = instr_ops(ops, put.data.data.format);
if (ops == NULL) {
snd_instr_unlock_ops(list);
goto __return;
}
len = ops->add_len;
if (put.data.type == SNDRV_SEQ_INSTR_ATYPE_ALIAS)
len = sizeof(struct snd_seq_instr);
instr = snd_seq_instr_new(len, atomic);
if (instr == NULL) {
snd_instr_unlock_ops(list);
result = -ENOMEM;
goto __return;
}
instr->ops = ops;
instr->instr = put.id.instr;
strlcpy(instr->name, put.data.name, sizeof(instr->name));
instr->type = put.data.type;
if (instr->type == SNDRV_SEQ_INSTR_ATYPE_DATA) {
result = ops->put(ops->private_data,
instr,
(void __user *)ev->data.ext.ptr + sizeof(struct snd_seq_instr_header),
ev->data.ext.len - sizeof(struct snd_seq_instr_header),
atomic,
put.cmd);
if (result < 0) {
snd_seq_instr_free(instr, atomic);
snd_instr_unlock_ops(list);
goto __return;
}
}
key = compute_hash_instr_key(&instr->instr);
spin_lock_irqsave(&list->lock, flags);
instr->next = list->hash[key];
list->hash[key] = instr;
list->count++;
spin_unlock_irqrestore(&list->lock, flags);
snd_instr_unlock_ops(list);
result = 0;
__return:
instr_result(ev, SNDRV_SEQ_EVENT_INSTR_PUT, result, atomic);
return result;
}
static int instr_get(struct snd_seq_kinstr_ops *ops,
struct snd_seq_kinstr_list *list,
struct snd_seq_event *ev,
int atomic, int hop)
{
return -ENXIO;
}
static int instr_free(struct snd_seq_kinstr_ops *ops,
struct snd_seq_kinstr_list *list,
struct snd_seq_event *ev,
int atomic, int hop)
{
struct snd_seq_instr_header ifree;
struct snd_seq_kinstr *instr, *prev;
int result = -EINVAL;
unsigned long flags;
unsigned int hash;
if ((ev->flags & SNDRV_SEQ_EVENT_LENGTH_MASK) != SNDRV_SEQ_EVENT_LENGTH_VARUSR)
goto __return;
if (ev->data.ext.len < sizeof(struct snd_seq_instr_header))
goto __return;
if (copy_from_user(&ifree, (void __user *)ev->data.ext.ptr,
sizeof(struct snd_seq_instr_header))) {
result = -EFAULT;
goto __return;
}
if (ifree.cmd == SNDRV_SEQ_INSTR_FREE_CMD_ALL ||
ifree.cmd == SNDRV_SEQ_INSTR_FREE_CMD_PRIVATE ||
ifree.cmd == SNDRV_SEQ_INSTR_FREE_CMD_CLUSTER) {
result = snd_seq_instr_list_free_cond(list, &ifree, ev->dest.client, atomic);
goto __return;
}
if (ifree.cmd == SNDRV_SEQ_INSTR_FREE_CMD_SINGLE) {
if (ifree.id.instr.std & 0xff000000) {
ifree.id.instr.std &= 0x00ffffff;
ifree.id.instr.std |= (unsigned int)ev->source.client << 24;
}
hash = compute_hash_instr_key(&ifree.id.instr);
snd_instr_lock_ops(list);
spin_lock_irqsave(&list->lock, flags);
instr = list->hash[hash];
prev = NULL;
while (instr) {
if (!compare_instr(&instr->instr, &ifree.id.instr, 1))
goto __free_single;
prev = instr;
instr = instr->next;
}
result = -ENOENT;
spin_unlock_irqrestore(&list->lock, flags);
snd_instr_unlock_ops(list);
goto __return;
__free_single:
if (prev) {
prev->next = instr->next;
} else {
list->hash[hash] = instr->next;
}
if (instr->ops && instr->ops->notify)
instr->ops->notify(instr->ops->private_data, instr,
SNDRV_SEQ_INSTR_NOTIFY_REMOVE);
while (instr->use) {
spin_unlock_irqrestore(&list->lock, flags);
schedule_timeout_uninterruptible(1);
spin_lock_irqsave(&list->lock, flags);
}
spin_unlock_irqrestore(&list->lock, flags);
result = snd_seq_instr_free(instr, atomic);
snd_instr_unlock_ops(list);
goto __return;
}
__return:
instr_result(ev, SNDRV_SEQ_EVENT_INSTR_FREE, result, atomic);
return result;
}
static int instr_list(struct snd_seq_kinstr_ops *ops,
struct snd_seq_kinstr_list *list,
struct snd_seq_event *ev,
int atomic, int hop)
{
return -ENXIO;
}
static int instr_cluster(struct snd_seq_kinstr_ops *ops,
struct snd_seq_kinstr_list *list,
struct snd_seq_event *ev,
int atomic, int hop)
{
return -ENXIO;
}
int snd_seq_instr_event(struct snd_seq_kinstr_ops *ops,
struct snd_seq_kinstr_list *list,
struct snd_seq_event *ev,
int client,
int atomic,
int hop)
{
int direct = 0;
snd_assert(ops != NULL && list != NULL && ev != NULL, return -EINVAL);
if (snd_seq_ev_is_direct(ev)) {
direct = 1;
switch (ev->type) {
case SNDRV_SEQ_EVENT_INSTR_BEGIN:
return instr_begin(ops, list, ev, atomic, hop);
case SNDRV_SEQ_EVENT_INSTR_END:
return instr_end(ops, list, ev, atomic, hop);
}
}
if ((list->flags & SNDRV_SEQ_INSTR_FLG_DIRECT) && !direct)
return -EINVAL;
switch (ev->type) {
case SNDRV_SEQ_EVENT_INSTR_INFO:
return instr_info(ops, list, ev, atomic, hop);
case SNDRV_SEQ_EVENT_INSTR_FINFO:
return instr_format_info(ops, list, ev, atomic, hop);
case SNDRV_SEQ_EVENT_INSTR_RESET:
return instr_reset(ops, list, ev, atomic, hop);
case SNDRV_SEQ_EVENT_INSTR_STATUS:
return instr_status(ops, list, ev, atomic, hop);
case SNDRV_SEQ_EVENT_INSTR_PUT:
return instr_put(ops, list, ev, atomic, hop);
case SNDRV_SEQ_EVENT_INSTR_GET:
return instr_get(ops, list, ev, atomic, hop);
case SNDRV_SEQ_EVENT_INSTR_FREE:
return instr_free(ops, list, ev, atomic, hop);
case SNDRV_SEQ_EVENT_INSTR_LIST:
return instr_list(ops, list, ev, atomic, hop);
case SNDRV_SEQ_EVENT_INSTR_CLUSTER:
return instr_cluster(ops, list, ev, atomic, hop);
}
return -EINVAL;
}
/*
* Init part
*/
static int __init alsa_seq_instr_init(void)
{
return 0;
}
static void __exit alsa_seq_instr_exit(void)
{
}
module_init(alsa_seq_instr_init)
module_exit(alsa_seq_instr_exit)
EXPORT_SYMBOL(snd_seq_instr_list_new);
EXPORT_SYMBOL(snd_seq_instr_list_free);
EXPORT_SYMBOL(snd_seq_instr_list_free_cond);
EXPORT_SYMBOL(snd_seq_instr_find);
EXPORT_SYMBOL(snd_seq_instr_free_use);
EXPORT_SYMBOL(snd_seq_instr_event);

1
sound/core/seq/seq_lock.c
View File

@ -19,7 +19,6 @@
*
*/
#include <sound/driver.h>
#include <sound/core.h>
#include "seq_lock.h"

Some files were not shown because too many files have changed in this diff Show More