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[alsa-devel] usr/include/linux/soundcard.h warnings for 'make headers_check'
by Jaswinder Singh Rajput 10 Feb '09

10 Feb '09

usr/include/linux/soundcard.h is giving following warnings for 'make headers_check': usr/include/linux/soundcard.h:1047: extern's make no sense in userspace usr/include/linux/soundcard.h:1048: extern's make no sense in userspace usr/include/linux/soundcard.h:1049: extern's make no sense in userspace usr/include/linux/soundcard.h:1050: extern's make no sense in userspace usr/include/linux/soundcard.h:1051: extern's make no sense in userspace usr/include/linux/soundcard.h:1053: extern's make no sense in userspace usr/include/linux/soundcard.h:1055: extern's make no sense in userspace usr/include/linux/soundcard.h:1056: extern's make no sense in userspace usr/include/linux/soundcard.h:1061: extern's make no sense in userspace usr/include/linux/soundcard.h:1062: extern's make no sense in userspace usr/include/linux/soundcard.h:1078: extern's make no sense in userspace usr/include/linux/soundcard.h:1079: extern's make no sense in userspace Do we need to fix these warnings. If yes, then how can we fix these warnings. Thanks -- JSR

5 19

[alsa-devel] [PATCH alsa-lib] Fix build error with -Werror=format-security
by Herton Ronaldo Krzesinski 09 Feb '09

09 Feb '09

src/alisp/alisp.c (F_princ): add missing format argument in call to snd_output_printf Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> --- src/alisp/alisp.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/src/alisp/alisp.c b/src/alisp/alisp.c index 4ad1d09..de3defb 100644 --- a/src/alisp/alisp.c +++ b/src/alisp/alisp.c @@ -1708,7 +1708,7 @@ static struct alisp_object * F_princ(struct alisp_instance *instance, struct ali delete_tree(instance, p1); p1 = eval(instance, car(p)); if (alisp_compare_type(p1, ALISP_OBJ_STRING)) - snd_output_printf(instance->out, p1->value.s); + snd_output_printf(instance->out, "%s", p1->value.s); else princ_object(instance->out, p1); n = cdr(p); -- 1.6.1.3

1 0

[alsa-devel] [topic-asoc 2.6.30][PATCH 1/2] ASoC: TLV320AIC3X: Fix volume ranges
by Jarkko Nikula 09 Feb '09

09 Feb '09

This is a minor fix but helps to define dB ranges for volume controls. Only DAC digital volume has full register value range from 0 to 127 but ADC PGA gain and output stage volume controls don't. For ADC PGA, maximum value is 119 and then it saturates to the same gain value of 59.5 dB. For output stages, value 117 corresponds to -78.3 dB and is muted for values 118 and above. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- sound/soc/codecs/tlv320aic3x.c | 38 +++++++++++++++++++------------------- 1 files changed, 19 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index ac73e69..cdb6dec 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -255,51 +255,51 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("PCM Playback Volume", LDAC_VOL, RDAC_VOL, 0, 0x7f, 1), SOC_DOUBLE_R("Line DAC Playback Volume", DACL1_2_LLOPM_VOL, - DACR1_2_RLOPM_VOL, 0, 0x7f, 1), + DACR1_2_RLOPM_VOL, 0, 118, 1), SOC_SINGLE("LineL Playback Switch", LLOPM_CTRL, 3, 0x01, 0), SOC_SINGLE("LineR Playback Switch", RLOPM_CTRL, 3, 0x01, 0), SOC_DOUBLE_R("LineL DAC Playback Volume", DACL1_2_LLOPM_VOL, - DACR1_2_LLOPM_VOL, 0, 0x7f, 1), + DACR1_2_LLOPM_VOL, 0, 118, 1), SOC_SINGLE("LineL Left PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL, - 0, 0x7f, 1), + 0, 118, 1), SOC_SINGLE("LineR Right PGA Bypass Playback Volume", PGAR_2_RLOPM_VOL, - 0, 0x7f, 1), + 0, 118, 1), SOC_DOUBLE_R("LineL Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL, - LINE2R_2_LLOPM_VOL, 0, 0x7f, 1), + LINE2R_2_LLOPM_VOL, 0, 118, 1), SOC_DOUBLE_R("LineR Line2 Bypass Playback Volume", LINE2L_2_RLOPM_VOL, - LINE2R_2_RLOPM_VOL, 0, 0x7f, 1), + LINE2R_2_RLOPM_VOL, 0, 118, 1), SOC_DOUBLE_R("Mono DAC Playback Volume", DACL1_2_MONOLOPM_VOL, - DACR1_2_MONOLOPM_VOL, 0, 0x7f, 1), + DACR1_2_MONOLOPM_VOL, 0, 118, 1), SOC_SINGLE("Mono DAC Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), SOC_DOUBLE_R("Mono PGA Bypass Playback Volume", PGAL_2_MONOLOPM_VOL, - PGAR_2_MONOLOPM_VOL, 0, 0x7f, 1), + PGAR_2_MONOLOPM_VOL, 0, 118, 1), SOC_DOUBLE_R("Mono Line2 Bypass Playback Volume", LINE2L_2_MONOLOPM_VOL, - LINE2R_2_MONOLOPM_VOL, 0, 0x7f, 1), + LINE2R_2_MONOLOPM_VOL, 0, 118, 1), SOC_DOUBLE_R("HP DAC Playback Volume", DACL1_2_HPLOUT_VOL, - DACR1_2_HPROUT_VOL, 0, 0x7f, 1), + DACR1_2_HPROUT_VOL, 0, 118, 1), SOC_DOUBLE_R("HP DAC Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3, 0x01, 0), SOC_DOUBLE_R("HP Right PGA Bypass Playback Volume", PGAR_2_HPLOUT_VOL, - PGAR_2_HPROUT_VOL, 0, 0x7f, 1), + PGAR_2_HPROUT_VOL, 0, 118, 1), SOC_SINGLE("HPL PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL, - 0, 0x7f, 1), + 0, 118, 1), SOC_SINGLE("HPR PGA Bypass Playback Volume", PGAL_2_HPROUT_VOL, - 0, 0x7f, 1), + 0, 118, 1), SOC_DOUBLE_R("HP Line2 Bypass Playback Volume", LINE2L_2_HPLOUT_VOL, - LINE2R_2_HPROUT_VOL, 0, 0x7f, 1), + LINE2R_2_HPROUT_VOL, 0, 118, 1), SOC_DOUBLE_R("HPCOM DAC Playback Volume", DACL1_2_HPLCOM_VOL, - DACR1_2_HPRCOM_VOL, 0, 0x7f, 1), + DACR1_2_HPRCOM_VOL, 0, 118, 1), SOC_DOUBLE_R("HPCOM DAC Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3, 0x01, 0), SOC_SINGLE("HPLCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL, - 0, 0x7f, 1), + 0, 118, 1), SOC_SINGLE("HPRCOM PGA Bypass Playback Volume", PGAL_2_HPRCOM_VOL, - 0, 0x7f, 1), + 0, 118, 1), SOC_DOUBLE_R("HPCOM Line2 Bypass Playback Volume", LINE2L_2_HPLCOM_VOL, - LINE2R_2_HPRCOM_VOL, 0, 0x7f, 1), + LINE2R_2_HPRCOM_VOL, 0, 118, 1), /* * Note: enable Automatic input Gain Controller with care. It can @@ -308,7 +308,7 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0), /* Input */ - SOC_DOUBLE_R("PGA Capture Volume", LADC_VOL, RADC_VOL, 0, 0x7f, 0), + SOC_DOUBLE_R("PGA Capture Volume", LADC_VOL, RADC_VOL, 0, 119, 0), SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1), SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), -- 1.5.6.5

2 2

[alsa-devel] capture hp dv3500
by Rodrigo Fresneda 09 Feb '09

09 Feb '09

The capture device for hp dv3500 (codec 92HD71B7X) does not work with alsa (1.0.19) There is no bug report for this, but maybe it has already been solved in newer version of alsa. Can anybody confirm? thanks, rodrigo.

2 5

[alsa-devel] No headphone with an Intel 82801I sound card
by Aurélien Croc 09 Feb '09

09 Feb '09

Hello, With the 2.6.27.10-0 (Mandriva) kernel driver, my sound card works quite well excepted the front digital microphone which didn't worked. Here is the ALSA script output: http://www.alsa-project.org/db/?f=31b5974fde70e1c5273f4248023481bb6766c46f I upgraded my kernel to the 2.6.27.10-1 and now the digital microphone works very well but the headphone output doesn't work. In other work, when i plug a speaker on it, the front integrated (to my laptop) speakers are muted but there is no sound in my external speaker. Here is the output of the ALSA script: http://www.alsa-project.org/db/?f=6def7ad107a215d22cef35a74d701216c5a496bd Thank you for your help. Regards, Aurélien

3 15

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 09 Feb '09

09 Feb '09

The following changes since commit 0664678a84c653bde844c7d91646259a25c6188b: Philipp Zabel (1): ASoC: pxa-ssp: fix SSP port request are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Eero Nurkkala (1): ASoC: TLV320AIC3X: Fix kcontrol's private value use in put callback Jarkko Nikula (1): ASoC: WM8990: Fix kcontrol's private value use in put callback Mark Brown (1): Merge branch 'for-2.6.29' into for-2.6.30 Mike Frysinger (2): ASoC: Blackfin: cleanup sport handling in ASoC Blackfin AC97 code ASoC: Blackfin: drop unnecessary dma casts Philipp Zabel (3): pxa/h5000: Setup I2S pins for pxa2xx-i2s pxa/spitz: Setup I2S pins for pxa2xx-i2s ASoC: pxa2xx-i2s: remove I2S pin setup Robert Jarzmik (1): ASoC: Add initial support of Mitac mioa701 device SoC. Roel Kluin (1): ASoC: count reaches 10001, not 10000. Timur Tabi (1): ASoC: optimize init sequence of Freescale MPC8610 sound drivers arch/arm/mach-pxa/h5000.c | 7 + arch/arm/mach-pxa/spitz.c | 6 + sound/soc/blackfin/bf5xx-ac97.c | 92 +++++--------- sound/soc/blackfin/bf5xx-ad73311.c | 4 +- sound/soc/blackfin/bf5xx-sport.c | 104 ++++++--------- sound/soc/codecs/tlv320aic3x.c | 11 +- sound/soc/codecs/wm8990.c | 4 +- sound/soc/fsl/fsl_dma.c | 178 ++++++++++++------------- sound/soc/fsl/fsl_ssi.c | 19 ++-- sound/soc/pxa/Kconfig | 9 ++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/mioa701_wm9713.c | 250 ++++++++++++++++++++++++++++++++++++ sound/soc/pxa/pxa2xx-i2s.c | 36 ----- 13 files changed, 456 insertions(+), 266 deletions(-) create mode 100644 sound/soc/pxa/mioa701_wm9713.c

2 1

[alsa-devel] ASoC updates for 2.6.29
by Mark Brown 09 Feb '09

09 Feb '09

The following changes since commit 64ca0404eed57f6c92290d55e949a7f46cbe4bf4: Liam Girdwood (1): ALSA: ASoC: email - update email addresses. are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.29 Eero Nurkkala (1): ASoC: TLV320AIC3X: Fix kcontrol's private value use in put callback Jarkko Nikula (1): ASoC: WM8990: Fix kcontrol's private value use in put callback sound/soc/codecs/tlv320aic3x.c | 11 +++++++---- sound/soc/codecs/wm8990.c | 4 +++- 2 files changed, 10 insertions(+), 5 deletions(-)

1 0

Re: [alsa-devel] HD-audio modularization
by Takashi Iwai 09 Feb '09

09 Feb '09

At Mon, 9 Feb 2009 09:39:34 +0100, Florian Mickler wrote: > > Hi! > > > For example, the recent SUSE init script has the following > > > > mod=$(grep -m1 -E '^(snd[^ ]*|ac97_bus) [0-9]+ 0' /proc/modules) > > while [ -n "$mod" ]; do > > mod=${mod%% *} > > /sbin/modprobe -r $mod > > mod=$(grep -m1 -E '^(snd[^ ]*|ac97_bus) [0-9]+ 0' /proc/modules) > > done > > > > But, basically it's stupid to unload modules at shtudown/reboot, just > > a waste of time. If possible, you should check the next runlevel and > > skip unloading if unnecessary. > > > > > > Takashi > > Yeah, it _is_ stupid to unload modules at shutdown. (and I kicked it > out of my init-script... but one wonders how it's supposed to work) > > modprobe -r snd_hda_codec_analog > gives me an error (''in use'') The codec driver is indeed used by the controller driver, snd-hda-intel. Takashi > # lsmod | grep snd_; modprobe -r snd_hda_codec_analog > snd_pcm_oss 42112 0 > snd_mixer_oss 16224 1 snd_pcm_oss > snd_hda_codec_analog 78144 1 > snd_hda_intel 28264 0 > snd_hda_codec 68992 2 snd_hda_codec_analog,snd_hda_intel > snd_pcm 86440 3 snd_pcm_oss,snd_hda_intel,snd_hda_codec > snd_timer 23472 1 snd_pcm > snd_page_alloc 10288 2 snd_hda_intel,snd_pcm > FATAL: Module snd_hda_codec_analog is in use. > > ...so, if i accidentially don't unload snd_hda_intel before > snd_hda_codec_analog it doesn't work? or are my tools somehow broken?

1 0

[alsa-devel] alsa driver api documentation
by Tim Blechmann 09 Feb '09

09 Feb '09

hello takashi, is it me, or is the alsa driver api documentation at [1] and [2] out of date? at least CONFIG_SND_DEBUG_DETECT seems to be renamed to CONFIG_SND_DEBUG_VERBOSE. would be great to see these changes in the api documentation ... thnx, tim [1] http://www.alsa-project.org/~tiwai/alsa-driver-api/ [2] http://www.kernel.org/pub/linux/kernel/people/tiwai/docs/alsa-driver-api/ -- tim(a)klingt.org http://tim.klingt.org I must say I find television very educational. The minute somebody turns it on, I go to the library and read a good book. Groucho Marx

2 1

[alsa-devel] HD-audio modularization
by Takashi Iwai 09 Feb '09

09 Feb '09

Hi, since my original diet plan for HD-audio driver isn't going well as expected, I decided the traditional strategy: divide and conquer. The below is the preliminary patch for modularizing snd-hda-intel. Now the codec-specific codes are split into several modules, and loaded automatically on demand. The patch is against the latest sound.git tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git It's also included in the recent sound-unstable tree, too (topic/hda-modularize branch): git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable-2.6.git A known issue is that "alsasound stop" may not work with this because of the explicit module refcount without dependency. But it's a script problem, basically. The similar problem must be there for snd-aoa, for example, too. Let me know if you see any other issues. thanks, Takashi === From 1289e9e8b42f973f2ab39e5f4f2239ff826c27e9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai <tiwai(a)suse.de> Date: Thu, 27 Nov 2008 15:47:11 +0100 Subject: [PATCH] ALSA: hda - Modularize HD-audio driver Split the monolithc HD-audio driver into several pieces: - snd-hda-intel HD-audio PCI controller driver; loaded via udev - snd-hda-codec HD-audio codec bus driver - snd-hda-codec-* Specific HD-audio codec drivers When built as modules, snd-hda-codec (that is invoked by snd-hda-intel) looks up the codec vendor ID and loads the corresponding codec module automatically via request_module(). When built in a kernel, each codec drivers are statically hooked up before probing the PCI. This patch adds appropriate EXPORT_SYMBOL_GPL()'s and the module information for each driver, and driver-linking codes between codec-bus and codec drivers. TODO: - Avoid EXPORT_SYMBOL*() when built-in kernel - Restore __devinit appropriately depending on the condition Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/Kconfig | 50 +++++++++++ sound/pci/hda/Makefile | 77 ++++++++++++---- sound/pci/hda/hda_beep.c | 2 + sound/pci/hda/hda_codec.c | 187 ++++++++++++++++++++++++++++++--------- sound/pci/hda/hda_codec.h | 11 +++ sound/pci/hda/hda_generic.c | 1 + sound/pci/hda/hda_hwdep.c | 2 +- sound/pci/hda/hda_patch.h | 24 ----- sound/pci/hda/patch_analog.c | 26 +++++- sound/pci/hda/patch_atihdmi.c | 31 ++++++- sound/pci/hda/patch_cmedia.c | 27 +++++- sound/pci/hda/patch_conexant.c | 28 ++++++- sound/pci/hda/patch_intelhdmi.c | 30 ++++++- sound/pci/hda/patch_nvhdmi.c | 27 +++++- sound/pci/hda/patch_realtek.c | 26 +++++- sound/pci/hda/patch_si3054.c | 35 +++++++- sound/pci/hda/patch_sigmatel.c | 27 +++++- sound/pci/hda/patch_via.c | 26 +++++- 18 files changed, 528 insertions(+), 109 deletions(-) delete mode 100644 sound/pci/hda/hda_patch.h diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 7c60f1a..717040a 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -46,6 +46,11 @@ config SND_HDA_CODEC_REALTEK Say Y here to include Realtek HD-audio codec support in snd-hda-intel driver, such as ALC880. + When the HD-audio driver is built as a module, the codec + support code is also built as another module, + snd-hda-codec-realtek. + This module is automatically loaded at probing. + config SND_HDA_CODEC_ANALOG bool "Build Analog Device HD-audio codec support" default y @@ -53,6 +58,11 @@ config SND_HDA_CODEC_ANALOG Say Y here to include Analog Device HD-audio codec support in snd-hda-intel driver, such as AD1986A. + When the HD-audio driver is built as a module, the codec + support code is also built as another module, + snd-hda-codec-analog. + This module is automatically loaded at probing. + config SND_HDA_CODEC_SIGMATEL bool "Build IDT/Sigmatel HD-audio codec support" default y @@ -60,6 +70,11 @@ config SND_HDA_CODEC_SIGMATEL Say Y here to include IDT (Sigmatel) HD-audio codec support in snd-hda-intel driver, such as STAC9200. + When the HD-audio driver is built as a module, the codec + support code is also built as another module, + snd-hda-codec-idt. + This module is automatically loaded at probing. + config SND_HDA_CODEC_VIA bool "Build VIA HD-audio codec support" default y @@ -67,6 +82,11 @@ config SND_HDA_CODEC_VIA Say Y here to include VIA HD-audio codec support in snd-hda-intel driver, such as VT1708. + When the HD-audio driver is built as a module, the codec + support code is also built as another module, + snd-hda-codec-via. + This module is automatically loaded at probing. + config SND_HDA_CODEC_ATIHDMI bool "Build ATI HDMI HD-audio codec support" default y @@ -74,6 +94,11 @@ config SND_HDA_CODEC_ATIHDMI Say Y here to include ATI HDMI HD-audio codec support in snd-hda-intel driver, such as ATI RS600 HDMI. + When the HD-audio driver is built as a module, the codec + support code is also built as another module, + snd-hda-codec-atihdmi. + This module is automatically loaded at probing. + config SND_HDA_CODEC_NVHDMI bool "Build NVIDIA HDMI HD-audio codec support" default y @@ -81,6 +106,11 @@ config SND_HDA_CODEC_NVHDMI Say Y here to include NVIDIA HDMI HD-audio codec support in snd-hda-intel driver, such as NVIDIA MCP78 HDMI. + When the HD-audio driver is built as a module, the codec + support code is also built as another module, + snd-hda-codec-nvhdmi. + This module is automatically loaded at probing. + config SND_HDA_CODEC_INTELHDMI bool "Build INTEL HDMI HD-audio codec support" default y @@ -88,6 +118,11 @@ config SND_HDA_CODEC_INTELHDMI Say Y here to include INTEL HDMI HD-audio codec support in snd-hda-intel driver, such as Eaglelake integrated HDMI. + When the HD-audio driver is built as a module, the codec + support code is also built as another module, + snd-hda-codec-intelhdmi. + This module is automatically loaded at probing. + config SND_HDA_ELD def_bool y depends on SND_HDA_CODEC_INTELHDMI @@ -99,6 +134,11 @@ config SND_HDA_CODEC_CONEXANT Say Y here to include Conexant HD-audio codec support in snd-hda-intel driver, such as CX20549. + When the HD-audio driver is built as a module, the codec + support code is also built as another module, + snd-hda-codec-conexant. + This module is automatically loaded at probing. + config SND_HDA_CODEC_CMEDIA bool "Build C-Media HD-audio codec support" default y @@ -106,6 +146,11 @@ config SND_HDA_CODEC_CMEDIA Say Y here to include C-Media HD-audio codec support in snd-hda-intel driver, such as CMI9880. + When the HD-audio driver is built as a module, the codec + support code is also built as another module, + snd-hda-codec-cmedia. + This module is automatically loaded at probing. + config SND_HDA_CODEC_SI3054 bool "Build Silicon Labs 3054 HD-modem codec support" default y @@ -113,6 +158,11 @@ config SND_HDA_CODEC_SI3054 Say Y here to include Silicon Labs 3054 HD-modem codec (and compatibles) support in snd-hda-intel driver. + When the HD-audio driver is built as a module, the codec + support code is also built as another module, + snd-hda-codec-si3054. + This module is automatically loaded at probing. + config SND_HDA_GENERIC bool "Enable generic HD-audio codec parser" default y diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 6daf5fd..50f9d09 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -1,22 +1,59 @@ -snd-hda-intel-y := hda_intel.o -# since snd-hda-intel is the only driver using hda-codec, -# merge it into a single module although it was originally -# designed to be individual modules -snd-hda-intel-y += hda_codec.o -snd-hda-intel-$(CONFIG_PROC_FS) += hda_proc.o -snd-hda-intel-$(CONFIG_SND_HDA_ELD) += hda_eld.o -snd-hda-intel-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o -snd-hda-intel-$(CONFIG_SND_HDA_INPUT_BEEP) += hda_beep.o -snd-hda-intel-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o -snd-hda-intel-$(CONFIG_SND_HDA_CODEC_REALTEK) += patch_realtek.o -snd-hda-intel-$(CONFIG_SND_HDA_CODEC_CMEDIA) += patch_cmedia.o -snd-hda-intel-$(CONFIG_SND_HDA_CODEC_ANALOG) += patch_analog.o -snd-hda-intel-$(CONFIG_SND_HDA_CODEC_SIGMATEL) += patch_sigmatel.o -snd-hda-intel-$(CONFIG_SND_HDA_CODEC_SI3054) += patch_si3054.o -snd-hda-intel-$(CONFIG_SND_HDA_CODEC_ATIHDMI) += patch_atihdmi.o -snd-hda-intel-$(CONFIG_SND_HDA_CODEC_CONEXANT) += patch_conexant.o -snd-hda-intel-$(CONFIG_SND_HDA_CODEC_VIA) += patch_via.o -snd-hda-intel-$(CONFIG_SND_HDA_CODEC_NVHDMI) += patch_nvhdmi.o -snd-hda-intel-$(CONFIG_SND_HDA_CODEC_INTELHDMI) += patch_intelhdmi.o +snd-hda-intel-objs := hda_intel.o +snd-hda-codec-y := hda_codec.o +snd-hda-codec-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o +snd-hda-codec-$(CONFIG_PROC_FS) += hda_proc.o +# snd-hda-codec-$(CONFIG_SND_HDA_ELD) += hda_eld.o +snd-hda-codec-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o +snd-hda-codec-$(CONFIG_SND_HDA_INPUT_BEEP) += hda_beep.o + +snd-hda-codec-realtek-objs := patch_realtek.o +snd-hda-codec-cmedia-objs := patch_cmedia.o +snd-hda-codec-analog-objs := patch_analog.o +snd-hda-codec-idt-objs := patch_sigmatel.o +snd-hda-codec-si3054-objs := patch_si3054.o +snd-hda-codec-atihdmi-objs := patch_atihdmi.o +snd-hda-codec-conexant-objs := patch_conexant.o +snd-hda-codec-via-objs := patch_via.o +snd-hda-codec-nvhdmi-objs := patch_nvhdmi.o +snd-hda-codec-intelhdmi-objs := patch_intelhdmi.o hda_eld.o + +# common driver +obj-$(CONFIG_SND_HDA_INTEL) := snd-hda-codec.o + +# codec drivers (note: CONFIG_SND_HDA_CODEC_XXX are booleans) +ifdef CONFIG_SND_HDA_CODEC_REALTEK +obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-realtek.o +endif +ifdef CONFIG_SND_HDA_CODEC_CMEDIA +obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-cmedia.o +endif +ifdef CONFIG_SND_HDA_CODEC_ANALOG +obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-analog.o +endif +ifdef CONFIG_SND_HDA_CODEC_SIGMATEL +obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-idt.o +endif +ifdef CONFIG_SND_HDA_CODEC_SI3054 +obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-si3054.o +endif +ifdef CONFIG_SND_HDA_CODEC_ATIHDMI +obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-atihdmi.o +endif +ifdef CONFIG_SND_HDA_CODEC_CONEXANT +obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-conexant.o +endif +ifdef CONFIG_SND_HDA_CODEC_VIA +obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-via.o +endif +ifdef CONFIG_SND_HDA_CODEC_NVHDMI +obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-nvhdmi.o +endif +ifdef CONFIG_SND_HDA_CODEC_INTELHDMI +obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-intelhdmi.o +endif + +# this must be the last entry after codec drivers; +# otherwise the codec patches won't be hooked before the PCI probe +# when built in kernel obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-intel.o diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 3ecd7e7..e6cc946 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -128,6 +128,7 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) INIT_WORK(&beep->beep_work, &snd_hda_generate_beep); return 0; } +EXPORT_SYMBOL_GPL(snd_hda_attach_beep_device); void snd_hda_detach_beep_device(struct hda_codec *codec) { @@ -140,3 +141,4 @@ void snd_hda_detach_beep_device(struct hda_codec *codec) kfree(beep); } } +EXPORT_SYMBOL_GPL(snd_hda_detach_beep_device); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 1cb85b7..4aa7b1b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -31,7 +31,6 @@ #include <sound/initval.h> #include "hda_local.h" #include <sound/hda_hwdep.h> -#include "hda_patch.h" /* codec presets */ /* * vendor / preset table @@ -62,39 +61,26 @@ static struct hda_vendor_id hda_vendor_ids[] = { {} /* terminator */ }; -static const struct hda_codec_preset *hda_preset_tables[] = { -#ifdef CONFIG_SND_HDA_CODEC_REALTEK - snd_hda_preset_realtek, -#endif -#ifdef CONFIG_SND_HDA_CODEC_CMEDIA - snd_hda_preset_cmedia, -#endif -#ifdef CONFIG_SND_HDA_CODEC_ANALOG - snd_hda_preset_analog, -#endif -#ifdef CONFIG_SND_HDA_CODEC_SIGMATEL - snd_hda_preset_sigmatel, -#endif -#ifdef CONFIG_SND_HDA_CODEC_SI3054 - snd_hda_preset_si3054, -#endif -#ifdef CONFIG_SND_HDA_CODEC_ATIHDMI - snd_hda_preset_atihdmi, -#endif -#ifdef CONFIG_SND_HDA_CODEC_CONEXANT - snd_hda_preset_conexant, -#endif -#ifdef CONFIG_SND_HDA_CODEC_VIA - snd_hda_preset_via, -#endif -#ifdef CONFIG_SND_HDA_CODEC_NVHDMI - snd_hda_preset_nvhdmi, -#endif -#ifdef CONFIG_SND_HDA_CODEC_INTELHDMI - snd_hda_preset_intelhdmi, -#endif - NULL -}; +static DEFINE_MUTEX(preset_mutex); +static LIST_HEAD(hda_preset_tables); + +int snd_hda_add_codec_preset(struct hda_codec_preset_list *preset) +{ + mutex_lock(&preset_mutex); + list_add_tail(&preset->list, &hda_preset_tables); + mutex_unlock(&preset_mutex); + return 0; +} +EXPORT_SYMBOL_GPL(snd_hda_add_codec_preset); + +int snd_hda_delete_codec_preset(struct hda_codec_preset_list *preset) +{ + mutex_lock(&preset_mutex); + list_del(&preset->list); + mutex_unlock(&preset_mutex); + return 0; +} +EXPORT_SYMBOL_GPL(snd_hda_delete_codec_preset); #ifdef CONFIG_SND_HDA_POWER_SAVE static void hda_power_work(struct work_struct *work); @@ -128,6 +114,7 @@ const char *snd_hda_get_jack_location(u32 cfg) } return "UNKNOWN"; } +EXPORT_SYMBOL_GPL(snd_hda_get_jack_location); const char *snd_hda_get_jack_connectivity(u32 cfg) { @@ -135,6 +122,7 @@ const char *snd_hda_get_jack_connectivity(u32 cfg) return jack_locations[(cfg >> (AC_DEFCFG_LOCATION_SHIFT + 4)) & 3]; } +EXPORT_SYMBOL_GPL(snd_hda_get_jack_connectivity); const char *snd_hda_get_jack_type(u32 cfg) { @@ -148,6 +136,7 @@ const char *snd_hda_get_jack_type(u32 cfg) return jack_types[(cfg & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT]; } +EXPORT_SYMBOL_GPL(snd_hda_get_jack_type); /* * Compose a 32bit command word to be sent to the HD-audio controller @@ -196,6 +185,7 @@ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, snd_hda_power_down(codec); return res; } +EXPORT_SYMBOL_GPL(snd_hda_codec_read); /** * snd_hda_codec_write - send a single command without waiting for response @@ -224,6 +214,7 @@ int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct, snd_hda_power_down(codec); return err; } +EXPORT_SYMBOL_GPL(snd_hda_codec_write); /** * snd_hda_sequence_write - sequence writes @@ -238,6 +229,7 @@ void snd_hda_sequence_write(struct hda_codec *codec, const struct hda_verb *seq) for (; seq->nid; seq++) snd_hda_codec_write(codec, seq->nid, 0, seq->verb, seq->param); } +EXPORT_SYMBOL_GPL(snd_hda_sequence_write); /** * snd_hda_get_sub_nodes - get the range of sub nodes @@ -259,6 +251,7 @@ int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid, *start_id = (parm >> 16) & 0x7fff; return (int)(parm & 0x7fff); } +EXPORT_SYMBOL_GPL(snd_hda_get_sub_nodes); /** * snd_hda_get_connections - get connection list @@ -347,6 +340,7 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, } return conns; } +EXPORT_SYMBOL_GPL(snd_hda_get_connections); /** @@ -381,6 +375,7 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex) return 0; } +EXPORT_SYMBOL_GPL(snd_hda_queue_unsol_event); /* * process queued unsolicited events @@ -482,7 +477,7 @@ static int snd_hda_bus_dev_register(struct snd_device *device) * * Returns 0 if successful, or a negative error code. */ -int __devinit snd_hda_bus_new(struct snd_card *card, +int /*__devinit*/ snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp, struct hda_bus **busp) { @@ -526,6 +521,7 @@ int __devinit snd_hda_bus_new(struct snd_card *card, *busp = bus; return 0; } +EXPORT_SYMBOL_GPL(snd_hda_bus_new); #ifdef CONFIG_SND_HDA_GENERIC #define is_generic_config(codec) \ @@ -534,19 +530,33 @@ int __devinit snd_hda_bus_new(struct snd_card *card, #define is_generic_config(codec) 0 #endif +#ifdef CONFIG_SND_HDA_INTEL_MODULE +#define HDA_MODREQ_MAX_COUNT 2 /* two request_modules()'s */ +#else +#define HDA_MODREQ_MAX_COUNT 0 +#endif + /* * find a matching codec preset */ static const struct hda_codec_preset * find_codec_preset(struct hda_codec *codec) { - const struct hda_codec_preset **tbl, *preset; + struct hda_codec_preset_list *tbl; + const struct hda_codec_preset *preset; + int mod_requested = 0; if (is_generic_config(codec)) return NULL; /* use the generic parser */ - for (tbl = hda_preset_tables; *tbl; tbl++) { - for (preset = *tbl; preset->id; preset++) { + again: + mutex_lock(&preset_mutex); + list_for_each_entry(tbl, &hda_preset_tables, list) { + if (!try_module_get(tbl->owner)) { + snd_printk(KERN_ERR "hda_codec: cannot module_get\n"); + continue; + } + for (preset = tbl->preset; preset->id; preset++) { u32 mask = preset->mask; if (preset->afg && preset->afg != codec->afg) continue; @@ -556,9 +566,27 @@ find_codec_preset(struct hda_codec *codec) mask = ~0; if (preset->id == (codec->vendor_id & mask) && (!preset->rev || - preset->rev == codec->revision_id)) + preset->rev == codec->revision_id)) { + mutex_unlock(&preset_mutex); + codec->owner = tbl->owner; return preset; + } } + module_put(tbl->owner); + } + mutex_unlock(&preset_mutex); + + if (mod_requested < HDA_MODREQ_MAX_COUNT) { + char name[32]; + if (!mod_requested) + snprintf(name, sizeof(name), "snd-hda-codec-id:%08x", + codec->vendor_id); + else + snprintf(name, sizeof(name), "snd-hda-codec-id:%04x*", + (codec->vendor_id >> 16) & 0xffff); + request_module(name); + mod_requested++; + goto again; } return NULL; } @@ -598,7 +626,7 @@ static int get_codec_name(struct hda_codec *codec) /* * look for an AFG and MFG nodes */ -static void __devinit setup_fg_nodes(struct hda_codec *codec) +static void /*__devinit*/ setup_fg_nodes(struct hda_codec *codec) { int i, total_nodes; hda_nid_t nid; @@ -661,6 +689,7 @@ static void snd_hda_codec_free(struct hda_codec *codec) codec->bus->caddr_tbl[codec->addr] = NULL; if (codec->patch_ops.free) codec->patch_ops.free(codec); + module_put(codec->owner); free_hda_cache(&codec->amp_cache); free_hda_cache(&codec->cmd_cache); kfree(codec->name); @@ -677,7 +706,7 @@ static void snd_hda_codec_free(struct hda_codec *codec) * * Returns 0 if successful, or a negative error code. */ -int __devinit snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, +int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, struct hda_codec **codecp) { struct hda_codec *codec; @@ -779,6 +808,7 @@ int __devinit snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, *codecp = codec; return 0; } +EXPORT_SYMBOL_GPL(snd_hda_codec_new); int snd_hda_codec_configure(struct hda_codec *codec) { @@ -838,6 +868,7 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, msleep(1); snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, format); } +EXPORT_SYMBOL_GPL(snd_hda_codec_setup_stream); void snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid) { @@ -851,6 +882,7 @@ void snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0); #endif } +EXPORT_SYMBOL_GPL(snd_hda_codec_cleanup_stream); /* * amp access functions @@ -862,7 +894,7 @@ void snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid) #define INFO_AMP_VOL(ch) (1 << (1 + (ch))) /* initialize the hash table */ -static void __devinit init_hda_cache(struct hda_cache_rec *cache, +static void /*__devinit*/ init_hda_cache(struct hda_cache_rec *cache, unsigned int record_size) { memset(cache, 0, sizeof(*cache)); @@ -932,6 +964,7 @@ u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction) } return info->amp_caps; } +EXPORT_SYMBOL_GPL(query_amp_caps); int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps) @@ -945,6 +978,7 @@ int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, info->head.val |= INFO_AMP_CAPS; return 0; } +EXPORT_SYMBOL_GPL(snd_hda_override_amp_caps); /* * read the current volume to info @@ -998,6 +1032,7 @@ int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch, return 0; return get_vol_mute(codec, info, nid, ch, direction, index); } +EXPORT_SYMBOL_GPL(snd_hda_codec_amp_read); /* * update the AMP value, mask = bit mask to set, val = the value @@ -1017,6 +1052,7 @@ int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, put_vol_mute(codec, info, nid, ch, direction, idx, val); return 1; } +EXPORT_SYMBOL_GPL(snd_hda_codec_amp_update); /* * update the AMP stereo with the same mask and value @@ -1030,6 +1066,7 @@ int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid, idx, mask, val); return ret; } +EXPORT_SYMBOL_GPL(snd_hda_codec_amp_stereo); #ifdef SND_HDA_NEEDS_RESUME /* resume the all amp commands from the cache */ @@ -1055,6 +1092,7 @@ void snd_hda_codec_resume_amp(struct hda_codec *codec) } } } +EXPORT_SYMBOL_GPL(snd_hda_codec_resume_amp); #endif /* SND_HDA_NEEDS_RESUME */ /* volume */ @@ -1082,6 +1120,7 @@ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol, uinfo->value.integer.max = caps; return 0; } +EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_volume_info); int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1101,6 +1140,7 @@ int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol, & HDA_AMP_VOLMASK; return 0; } +EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_volume_get); int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1125,6 +1165,7 @@ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol, snd_hda_power_down(codec); return change; } +EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_volume_put); int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *_tlv) @@ -1151,6 +1192,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, return -EFAULT; return 0; } +EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_tlv); /* * set (static) TLV for virtual master volume; recalculated as max 0dB @@ -1170,6 +1212,7 @@ void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir, tlv[2] = -nums * step; tlv[3] = step; } +EXPORT_SYMBOL_GPL(snd_hda_set_vmaster_tlv); /* find a mixer control element with the given name */ static struct snd_kcontrol * @@ -1189,6 +1232,7 @@ struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, { return _snd_hda_find_mixer_ctl(codec, name, 0); } +EXPORT_SYMBOL_GPL(snd_hda_find_mixer_ctl); /* Add a control element and assign to the codec */ int snd_hda_ctl_add(struct hda_codec *codec, struct snd_kcontrol *kctl) @@ -1205,6 +1249,7 @@ int snd_hda_ctl_add(struct hda_codec *codec, struct snd_kcontrol *kctl) *knewp = kctl; return 0; } +EXPORT_SYMBOL_GPL(snd_hda_ctl_add); #ifdef CONFIG_SND_HDA_RECONFIG /* Clear all controls assigned to the given codec */ @@ -1243,6 +1288,8 @@ void snd_hda_codec_reset(struct hda_codec *codec) codec->num_pcms = 0; codec->pcm_info = NULL; codec->preset = NULL; + module_put(codec->owner); + codec->owner = NULL; } #endif /* CONFIG_SND_HDA_RECONFIG */ @@ -1281,6 +1328,7 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, } return 0; } +EXPORT_SYMBOL_GPL(snd_hda_add_vmaster); /* switch */ int snd_hda_mixer_amp_switch_info(struct snd_kcontrol *kcontrol, @@ -1294,6 +1342,7 @@ int snd_hda_mixer_amp_switch_info(struct snd_kcontrol *kcontrol, uinfo->value.integer.max = 1; return 0; } +EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_switch_info); int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1313,6 +1362,7 @@ int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol, HDA_AMP_MUTE) ? 0 : 1; return 0; } +EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_switch_get); int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1343,6 +1393,7 @@ int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, snd_hda_power_down(codec); return change; } +EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_switch_put); /* * bound volume controls @@ -1368,6 +1419,7 @@ int snd_hda_mixer_bind_switch_get(struct snd_kcontrol *kcontrol, mutex_unlock(&codec->spdif_mutex); return err; } +EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_switch_get); int snd_hda_mixer_bind_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1391,6 +1443,7 @@ int snd_hda_mixer_bind_switch_put(struct snd_kcontrol *kcontrol, mutex_unlock(&codec->spdif_mutex); return err < 0 ? err : change; } +EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_switch_put); /* * generic bound volume/swtich controls @@ -1410,6 +1463,7 @@ int snd_hda_mixer_bind_ctls_info(struct snd_kcontrol *kcontrol, mutex_unlock(&codec->spdif_mutex); return err; } +EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_ctls_info); int snd_hda_mixer_bind_ctls_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1426,6 +1480,7 @@ int snd_hda_mixer_bind_ctls_get(struct snd_kcontrol *kcontrol, mutex_unlock(&codec->spdif_mutex); return err; } +EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_ctls_get); int snd_hda_mixer_bind_ctls_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1448,6 +1503,7 @@ int snd_hda_mixer_bind_ctls_put(struct snd_kcontrol *kcontrol, mutex_unlock(&codec->spdif_mutex); return err < 0 ? err : change; } +EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_ctls_put); int snd_hda_mixer_bind_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *tlv) @@ -1464,6 +1520,7 @@ int snd_hda_mixer_bind_tlv(struct snd_kcontrol *kcontrol, int op_flag, mutex_unlock(&codec->spdif_mutex); return err; } +EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_tlv); struct hda_ctl_ops snd_hda_bind_vol = { .info = snd_hda_mixer_amp_volume_info, @@ -1471,6 +1528,7 @@ struct hda_ctl_ops snd_hda_bind_vol = { .put = snd_hda_mixer_amp_volume_put, .tlv = snd_hda_mixer_amp_tlv }; +EXPORT_SYMBOL_GPL(snd_hda_bind_vol); struct hda_ctl_ops snd_hda_bind_sw = { .info = snd_hda_mixer_amp_switch_info, @@ -1478,6 +1536,7 @@ struct hda_ctl_ops snd_hda_bind_sw = { .put = snd_hda_mixer_amp_switch_put, .tlv = snd_hda_mixer_amp_tlv }; +EXPORT_SYMBOL_GPL(snd_hda_bind_sw); /* * SPDIF out controls @@ -1739,6 +1798,7 @@ int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid) codec->spdif_status = convert_to_spdif_status(codec->spdif_ctls); return 0; } +EXPORT_SYMBOL_GPL(snd_hda_create_spdif_out_ctls); /* * SPDIF sharing with analog output @@ -1776,6 +1836,7 @@ int snd_hda_create_spdif_share_sw(struct hda_codec *codec, return snd_hda_ctl_add(codec, snd_ctl_new1(&spdif_share_sw, mout)); } +EXPORT_SYMBOL_GPL(snd_hda_create_spdif_share_sw); /* * SPDIF input @@ -1885,6 +1946,7 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid) AC_DIG1_ENABLE; return 0; } +EXPORT_SYMBOL_GPL(snd_hda_create_spdif_in_ctls); #ifdef SND_HDA_NEEDS_RESUME /* @@ -1930,6 +1992,7 @@ int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, snd_hda_power_down(codec); return err; } +EXPORT_SYMBOL_GPL(snd_hda_codec_write_cache); /* resume the all commands from the cache */ void snd_hda_codec_resume_cache(struct hda_codec *codec) @@ -1945,6 +2008,7 @@ void snd_hda_codec_resume_cache(struct hda_codec *codec) get_cmd_cache_cmd(key), buffer->val); } } +EXPORT_SYMBOL_GPL(snd_hda_codec_resume_cache); /** * snd_hda_sequence_write_cache - sequence writes with caching @@ -1962,6 +2026,7 @@ void snd_hda_sequence_write_cache(struct hda_codec *codec, snd_hda_codec_write_cache(codec, seq->nid, 0, seq->verb, seq->param); } +EXPORT_SYMBOL_GPL(snd_hda_sequence_write_cache); #endif /* SND_HDA_NEEDS_RESUME */ /* @@ -2080,7 +2145,7 @@ static void hda_call_codec_resume(struct hda_codec *codec) * * Returns 0 if successful, otherwise a negative error code. */ -int __devinit snd_hda_build_controls(struct hda_bus *bus) +int /*__devinit*/ snd_hda_build_controls(struct hda_bus *bus) { struct hda_codec *codec; @@ -2091,6 +2156,7 @@ int __devinit snd_hda_build_controls(struct hda_bus *bus) } return 0; } +EXPORT_SYMBOL_GPL(snd_hda_build_controls); int snd_hda_codec_build_controls(struct hda_codec *codec) { @@ -2202,6 +2268,7 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate, return val; } +EXPORT_SYMBOL_GPL(snd_hda_calc_stream_format); /** * snd_hda_query_supported_pcm - query the supported PCM rates and formats @@ -2381,6 +2448,7 @@ int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid, return 1; } +EXPORT_SYMBOL_GPL(snd_hda_is_supported_format); /* * PCM stuff @@ -2578,6 +2646,7 @@ int __devinit snd_hda_build_pcms(struct hda_bus *bus) } return 0; } +EXPORT_SYMBOL_GPL(snd_hda_build_pcms); /** * snd_hda_check_board_config - compare the current codec with the config table @@ -2633,6 +2702,7 @@ int snd_hda_check_board_config(struct hda_codec *codec, } return -1; } +EXPORT_SYMBOL_GPL(snd_hda_check_board_config); /** * snd_hda_add_new_ctls - create controls from the array @@ -2668,6 +2738,7 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew) } return 0; } +EXPORT_SYMBOL_GPL(snd_hda_add_new_ctls); #ifdef CONFIG_SND_HDA_POWER_SAVE static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, @@ -2710,6 +2781,10 @@ void snd_hda_power_up(struct hda_codec *codec) cancel_delayed_work(&codec->power_work); codec->power_transition = 0; } +EXPORT_SYMBOL_GPL(snd_hda_power_up); + +#define power_save(codec) \ + ((codec)->bus->power_save ? *(codec)->bus->power_save : 0) #define power_save(codec) \ ((codec)->bus->power_save ? *(codec)->bus->power_save : 0) @@ -2725,6 +2800,7 @@ void snd_hda_power_down(struct hda_codec *codec) msecs_to_jiffies(power_save(codec) * 1000)); } } +EXPORT_SYMBOL_GPL(snd_hda_power_down); int snd_hda_check_amp_list_power(struct hda_codec *codec, struct hda_loopback_check *check, @@ -2761,6 +2837,7 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, } return 0; } +EXPORT_SYMBOL_GPL(snd_hda_check_amp_list_power); #endif /* @@ -2780,6 +2857,7 @@ int snd_hda_ch_mode_info(struct hda_codec *codec, chmode[uinfo->value.enumerated.item].channels); return 0; } +EXPORT_SYMBOL_GPL(snd_hda_ch_mode_info); int snd_hda_ch_mode_get(struct hda_codec *codec, struct snd_ctl_elem_value *ucontrol, @@ -2797,6 +2875,7 @@ int snd_hda_ch_mode_get(struct hda_codec *codec, } return 0; } +EXPORT_SYMBOL_GPL(snd_hda_ch_mode_get); int snd_hda_ch_mode_put(struct hda_codec *codec, struct snd_ctl_elem_value *ucontrol, @@ -2817,6 +2896,7 @@ int snd_hda_ch_mode_put(struct hda_codec *codec, snd_hda_sequence_write_cache(codec, chmode[mode].sequence); return 1; } +EXPORT_SYMBOL_GPL(snd_hda_ch_mode_put); /* * input MUX helper @@ -2837,6 +2917,7 @@ int snd_hda_input_mux_info(const struct hda_input_mux *imux, strcpy(uinfo->value.enumerated.name, imux->items[index].label); return 0; } +EXPORT_SYMBOL_GPL(snd_hda_input_mux_info); int snd_hda_input_mux_put(struct hda_codec *codec, const struct hda_input_mux *imux, @@ -2858,6 +2939,7 @@ int snd_hda_input_mux_put(struct hda_codec *codec, *cur_val = idx; return 1; } +EXPORT_SYMBOL_GPL(snd_hda_input_mux_put); /* @@ -2910,6 +2992,7 @@ int snd_hda_multi_out_dig_open(struct hda_codec *codec, mutex_unlock(&codec->spdif_mutex); return 0; } +EXPORT_SYMBOL_GPL(snd_hda_multi_out_dig_open); int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, struct hda_multi_out *mout, @@ -2922,6 +3005,7 @@ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, mutex_unlock(&codec->spdif_mutex); return 0; } +EXPORT_SYMBOL_GPL(snd_hda_multi_out_dig_prepare); /* * release the digital out @@ -2934,6 +3018,7 @@ int snd_hda_multi_out_dig_close(struct hda_codec *codec, mutex_unlock(&codec->spdif_mutex); return 0; } +EXPORT_SYMBOL_GPL(snd_hda_multi_out_dig_close); /* * set up more restrictions for analog out @@ -2973,6 +3058,7 @@ int snd_hda_multi_out_analog_open(struct hda_codec *codec, return snd_pcm_hw_constraint_step(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, 2); } +EXPORT_SYMBOL_GPL(snd_hda_multi_out_analog_open); /* * set up the i/o for analog out @@ -3031,6 +3117,7 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, } return 0; } +EXPORT_SYMBOL_GPL(snd_hda_multi_out_analog_prepare); /* * clean up the setting for analog out @@ -3057,6 +3144,7 @@ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, mutex_unlock(&codec->spdif_mutex); return 0; } +EXPORT_SYMBOL_GPL(snd_hda_multi_out_analog_cleanup); /* * Helper for automatic pin configuration @@ -3342,11 +3430,13 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, return 0; } +EXPORT_SYMBOL_GPL(snd_hda_parse_pin_def_config); /* labels for input pins */ const char *auto_pin_cfg_labels[AUTO_PIN_LAST] = { "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux" }; +EXPORT_SYMBOL_GPL(auto_pin_cfg_labels); #ifdef CONFIG_PM @@ -3374,6 +3464,7 @@ int snd_hda_suspend(struct hda_bus *bus, pm_message_t state) } return 0; } +EXPORT_SYMBOL_GPL(snd_hda_suspend); /** * snd_hda_resume - resume the codecs @@ -3394,7 +3485,8 @@ int snd_hda_resume(struct hda_bus *bus) } return 0; } -#endif +EXPORT_SYMBOL_GPL(snd_hda_resume); +#endif /* CONFIG_PM */ /* * generic arrays @@ -3423,6 +3515,7 @@ void *snd_array_new(struct snd_array *array) } return snd_array_elem(array, array->used++); } +EXPORT_SYMBOL_GPL(snd_array_new); /* free the given array elements */ void snd_array_free(struct snd_array *array) @@ -3432,6 +3525,7 @@ void snd_array_free(struct snd_array *array) array->alloced = 0; array->list = NULL; } +EXPORT_SYMBOL_GPL(snd_array_free); /* * used by hda_proc.c and hda_eld.c @@ -3450,6 +3544,7 @@ void snd_print_pcm_rates(int pcm, char *buf, int buflen) buf[j] = '\0'; /* necessary when j == 0 */ } +EXPORT_SYMBOL_GPL(snd_print_pcm_rates); void snd_print_pcm_bits(int pcm, char *buf, int buflen) { @@ -3462,3 +3557,7 @@ void snd_print_pcm_bits(int pcm, char *buf, int buflen) buf[j] = '\0'; /* necessary when j == 0 */ } +EXPORT_SYMBOL_GPL(snd_print_pcm_bits); + +MODULE_DESCRIPTION("HDA codec core"); +MODULE_LICENSE("GPL"); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 9fe0b67..586ea08 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -642,6 +642,16 @@ struct hda_codec_preset { int (*patch)(struct hda_codec *codec); }; +struct hda_codec_preset_list { + const struct hda_codec_preset *preset; + struct module *owner; + struct list_head list; +}; + +/* initial hook */ +int snd_hda_add_codec_preset(struct hda_codec_preset_list *preset); +int snd_hda_delete_codec_preset(struct hda_codec_preset_list *preset); + /* ops set by the preset patch */ struct hda_codec_ops { int (*build_controls)(struct hda_codec *codec); @@ -735,6 +745,7 @@ struct hda_codec { /* detected preset */ const struct hda_codec_preset *preset; + struct module *owner; const char *name; /* codec name */ const char *modelname; /* model name for preset */ diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 98ff010..65745e9 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -1101,3 +1101,4 @@ int snd_hda_parse_generic_codec(struct hda_codec *codec) snd_hda_generic_free(codec); return err; } +EXPORT_SYMBOL(snd_hda_parse_generic_codec); diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 173af48..300ab40 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -116,7 +116,7 @@ static void hwdep_free(struct snd_hwdep *hwdep) clear_hwdep_elements(hwdep->private_data); } -int __devinit snd_hda_create_hwdep(struct hda_codec *codec) +int /*__devinit*/ snd_hda_create_hwdep(struct hda_codec *codec) { char hwname[16]; struct snd_hwdep *hwdep; diff --git a/sound/pci/hda/hda_patch.h b/sound/pci/hda/hda_patch.h deleted file mode 100644 index 38212c1..0000000 --- a/sound/pci/hda/hda_patch.h +++ /dev/null @@ -1,24 +0,0 @@ -/* - * HDA Patches - included by hda_codec.c - */ - -/* Realtek codecs */ -extern struct hda_codec_preset snd_hda_preset_realtek[]; -/* C-Media codecs */ -extern struct hda_codec_preset snd_hda_preset_cmedia[]; -/* Analog Devices codecs */ -extern struct hda_codec_preset snd_hda_preset_analog[]; -/* SigmaTel codecs */ -extern struct hda_codec_preset snd_hda_preset_sigmatel[]; -/* SiLabs 3054/3055 modem codecs */ -extern struct hda_codec_preset snd_hda_preset_si3054[]; -/* ATI HDMI codecs */ -extern struct hda_codec_preset snd_hda_preset_atihdmi[]; -/* Conexant audio codec */ -extern struct hda_codec_preset snd_hda_preset_conexant[]; -/* VIA codecs */ -extern struct hda_codec_preset snd_hda_preset_via[]; -/* NVIDIA HDMI codecs */ -extern struct hda_codec_preset snd_hda_preset_nvhdmi[]; -/* INTEL HDMI codecs */ -extern struct hda_codec_preset snd_hda_preset_intelhdmi[]; diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 0cc6be1..c1918a1 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -27,7 +27,6 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" -#include "hda_patch.h" struct ad198x_spec { struct snd_kcontrol_new *mixers[5]; @@ -4308,7 +4307,7 @@ static int patch_ad1882(struct hda_codec *codec) /* * patch entries */ -struct hda_codec_preset snd_hda_preset_analog[] = { +static struct hda_codec_preset snd_hda_preset_analog[] = { { .id = 0x11d4184a, .name = "AD1884A", .patch = patch_ad1884a }, { .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 }, { .id = 0x11d41883, .name = "AD1883", .patch = patch_ad1884a }, @@ -4326,3 +4325,26 @@ struct hda_codec_preset snd_hda_preset_analog[] = { { .id = 0x11d4989b, .name = "AD1989B", .patch = patch_ad1988 }, {} /* terminator */ }; + +MODULE_ALIAS("snd-hda-codec-id:11d4*"); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Analog Devices HD-audio codec"); + +static struct hda_codec_preset_list analog_list = { + .preset = snd_hda_preset_analog, + .owner = THIS_MODULE, +}; + +static int __init patch_analog_init(void) +{ + return snd_hda_add_codec_preset(&analog_list); +} + +static void __exit patch_analog_exit(void) +{ + snd_hda_delete_codec_preset(&analog_list); +} + +module_init(patch_analog_init) +module_exit(patch_analog_exit) diff --git a/sound/pci/hda/patch_atihdmi.c b/sound/pci/hda/patch_atihdmi.c index 5603a1a..5887b82 100644 --- a/sound/pci/hda/patch_atihdmi.c +++ b/sound/pci/hda/patch_atihdmi.c @@ -27,7 +27,6 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" -#include "hda_patch.h" struct atihdmi_spec { struct hda_multi_out multiout; @@ -187,7 +186,7 @@ static int patch_atihdmi(struct hda_codec *codec) /* * patch entries */ -struct hda_codec_preset snd_hda_preset_atihdmi[] = { +static struct hda_codec_preset snd_hda_preset_atihdmi[] = { { .id = 0x1002793c, .name = "ATI RS600 HDMI", .patch = patch_atihdmi }, { .id = 0x10027919, .name = "ATI RS600 HDMI", .patch = patch_atihdmi }, { .id = 0x1002791a, .name = "ATI RS690/780 HDMI", .patch = patch_atihdmi }, @@ -196,3 +195,31 @@ struct hda_codec_preset snd_hda_preset_atihdmi[] = { { .id = 0x17e80047, .name = "Chrontel HDMI", .patch = patch_atihdmi }, {} /* terminator */ }; + +MODULE_ALIAS("snd-hda-codec-id:1002793c"); +MODULE_ALIAS("snd-hda-codec-id:10027919"); +MODULE_ALIAS("snd-hda-codec-id:1002791a"); +MODULE_ALIAS("snd-hda-codec-id:1002aa01"); +MODULE_ALIAS("snd-hda-codec-id:10951390"); +MODULE_ALIAS("snd-hda-codec-id:17e80047"); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("ATI HDMI HD-audio codec"); + +static struct hda_codec_preset_list atihdmi_list = { + .preset = snd_hda_preset_atihdmi, + .owner = THIS_MODULE, +}; + +static int __init patch_atihdmi_init(void) +{ + return snd_hda_add_codec_preset(&atihdmi_list); +} + +static void __exit patch_atihdmi_exit(void) +{ + snd_hda_delete_codec_preset(&atihdmi_list); +} + +module_init(patch_atihdmi_init) +module_exit(patch_atihdmi_exit) diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 6ef57fb..f3ebe83 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -28,7 +28,6 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" -#include "hda_patch.h" #define NUM_PINS 11 @@ -736,8 +735,32 @@ static int patch_cmi9880(struct hda_codec *codec) /* * patch entries */ -struct hda_codec_preset snd_hda_preset_cmedia[] = { +static struct hda_codec_preset snd_hda_preset_cmedia[] = { { .id = 0x13f69880, .name = "CMI9880", .patch = patch_cmi9880 }, { .id = 0x434d4980, .name = "CMI9880", .patch = patch_cmi9880 }, {} /* terminator */ }; + +MODULE_ALIAS("snd-hda-codec-id:13f69880"); +MODULE_ALIAS("snd-hda-codec-id:434d4980"); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("C-Media HD-audio codec"); + +static struct hda_codec_preset_list cmedia_list = { + .preset = snd_hda_preset_cmedia, + .owner = THIS_MODULE, +}; + +static int __init patch_cmedia_init(void) +{ + return snd_hda_add_codec_preset(&cmedia_list); +} + +static void __exit patch_cmedia_exit(void) +{ + snd_hda_delete_codec_preset(&cmedia_list); +} + +module_init(patch_cmedia_init) +module_exit(patch_cmedia_exit) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 0760107..b20e1ce 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -27,7 +27,6 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" -#include "hda_patch.h" #define CXT_PIN_DIR_IN 0x00 #define CXT_PIN_DIR_OUT 0x01 @@ -1771,7 +1770,7 @@ static int patch_cxt5051(struct hda_codec *codec) /* */ -struct hda_codec_preset snd_hda_preset_conexant[] = { +static struct hda_codec_preset snd_hda_preset_conexant[] = { { .id = 0x14f15045, .name = "CX20549 (Venice)", .patch = patch_cxt5045 }, { .id = 0x14f15047, .name = "CX20551 (Waikiki)", @@ -1780,3 +1779,28 @@ struct hda_codec_preset snd_hda_preset_conexant[] = { .patch = patch_cxt5051 }, {} /* terminator */ }; + +MODULE_ALIAS("snd-hda-codec-id:14f15045"); +MODULE_ALIAS("snd-hda-codec-id:14f15047"); +MODULE_ALIAS("snd-hda-codec-id:14f15051"); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Conexant HD-audio codec"); + +static struct hda_codec_preset_list conexant_list = { + .preset = snd_hda_preset_conexant, + .owner = THIS_MODULE, +}; + +static int __init patch_conexant_init(void) +{ + return snd_hda_add_codec_preset(&conexant_list); +} + +static void __exit patch_conexant_exit(void) +{ + snd_hda_delete_codec_preset(&conexant_list); +} + +module_init(patch_conexant_init) +module_exit(patch_conexant_exit) diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index fe08bef..290da56 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -32,7 +32,6 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" -#include "hda_patch.h" #define CVT_NID 0x02 /* audio converter */ #define PIN_NID 0x03 /* HDMI output pin */ @@ -675,7 +674,7 @@ static int patch_intel_hdmi(struct hda_codec *codec) return 0; } -struct hda_codec_preset snd_hda_preset_intelhdmi[] = { +static struct hda_codec_preset snd_hda_preset_intelhdmi[] = { { .id = 0x808629fb, .name = "INTEL G45 DEVCL", .patch = patch_intel_hdmi }, { .id = 0x80862801, .name = "INTEL G45 DEVBLC", .patch = patch_intel_hdmi }, { .id = 0x80862802, .name = "INTEL G45 DEVCTG", .patch = patch_intel_hdmi }, @@ -683,3 +682,30 @@ struct hda_codec_preset snd_hda_preset_intelhdmi[] = { { .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_intel_hdmi }, {} /* terminator */ }; + +MODULE_ALIAS("snd-hda-codec-id:808629fb"); +MODULE_ALIAS("snd-hda-codec-id:80862801"); +MODULE_ALIAS("snd-hda-codec-id:80862802"); +MODULE_ALIAS("snd-hda-codec-id:80862803"); +MODULE_ALIAS("snd-hda-codec-id:10951392"); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Intel HDMI HD-audio codec"); + +static struct hda_codec_preset_list intel_list = { + .preset = snd_hda_preset_intelhdmi, + .owner = THIS_MODULE, +}; + +static int __init patch_intelhdmi_init(void) +{ + return snd_hda_add_codec_preset(&intel_list); +} + +static void __exit patch_intelhdmi_exit(void) +{ + snd_hda_delete_codec_preset(&intel_list); +} + +module_init(patch_intelhdmi_init) +module_exit(patch_intelhdmi_exit) diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index 1360d54..e23de55 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -28,7 +28,6 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" -#include "hda_patch.h" struct nvhdmi_spec { struct hda_multi_out multiout; @@ -159,8 +158,32 @@ static int patch_nvhdmi(struct hda_codec *codec) /* * patch entries */ -struct hda_codec_preset snd_hda_preset_nvhdmi[] = { +static struct hda_codec_preset snd_hda_preset_nvhdmi[] = { { .id = 0x10de0002, .name = "NVIDIA MCP78 HDMI", .patch = patch_nvhdmi }, { .id = 0x10de0007, .name = "NVIDIA MCP7A HDMI", .patch = patch_nvhdmi }, {} /* terminator */ }; + +MODULE_ALIAS("snd-hda-codec-id:10de0002"); +MODULE_ALIAS("snd-hda-codec-id:10de0007"); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Nvidia HDMI HD-audio codec"); + +static struct hda_codec_preset_list nvhdmi_list = { + .preset = snd_hda_preset_nvhdmi, + .owner = THIS_MODULE, +}; + +static int __init patch_nvhdmi_init(void) +{ + return snd_hda_add_codec_preset(&nvhdmi_list); +} + +static void __exit patch_nvhdmi_exit(void) +{ + snd_hda_delete_codec_preset(&nvhdmi_list); +} + +module_init(patch_nvhdmi_init) +module_exit(patch_nvhdmi_exit) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9cd2545..ba640d3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -30,7 +30,6 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" -#include "hda_patch.h" #define ALC880_FRONT_EVENT 0x01 #define ALC880_DCVOL_EVENT 0x02 @@ -16579,7 +16578,7 @@ static int patch_alc662(struct hda_codec *codec) /* * patch entries */ -struct hda_codec_preset snd_hda_preset_realtek[] = { +static struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 }, { .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 }, { .id = 0x10ec0267, .name = "ALC267", .patch = patch_alc268 }, @@ -16611,3 +16610,26 @@ struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc883 }, {} /* terminator */ }; + +MODULE_ALIAS("snd-hda-codec-id:10ec*"); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Realtek HD-audio codec"); + +static struct hda_codec_preset_list realtek_list = { + .preset = snd_hda_preset_realtek, + .owner = THIS_MODULE, +}; + +static int __init patch_realtek_init(void) +{ + return snd_hda_add_codec_preset(&realtek_list); +} + +static void __exit patch_realtek_exit(void) +{ + snd_hda_delete_codec_preset(&realtek_list); +} + +module_init(patch_realtek_init) +module_exit(patch_realtek_exit) diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index 9332b63..43b436c 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -28,7 +28,6 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" -#include "hda_patch.h" /* si3054 verbs */ #define SI3054_VERB_READ_NODE 0x900 @@ -283,7 +282,7 @@ static int patch_si3054(struct hda_codec *codec) /* * patch entries */ -struct hda_codec_preset snd_hda_preset_si3054[] = { +static struct hda_codec_preset snd_hda_preset_si3054[] = { { .id = 0x163c3055, .name = "Si3054", .patch = patch_si3054 }, { .id = 0x163c3155, .name = "Si3054", .patch = patch_si3054 }, { .id = 0x11c13026, .name = "Si3054", .patch = patch_si3054 }, @@ -301,3 +300,35 @@ struct hda_codec_preset snd_hda_preset_si3054[] = { {} }; +MODULE_ALIAS("snd-hda-codec-id:163c3055"); +MODULE_ALIAS("snd-hda-codec-id:163c3155"); +MODULE_ALIAS("snd-hda-codec-id:11c13026"); +MODULE_ALIAS("snd-hda-codec-id:11c13055"); +MODULE_ALIAS("snd-hda-codec-id:11c13155"); +MODULE_ALIAS("snd-hda-codec-id:10573055"); +MODULE_ALIAS("snd-hda-codec-id:10573057"); +MODULE_ALIAS("snd-hda-codec-id:10573155"); +MODULE_ALIAS("snd-hda-codec-id:11063288"); +MODULE_ALIAS("snd-hda-codec-id:15433155"); +MODULE_ALIAS("snd-hda-codec-id:18540018"); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Si3054 HD-audio modem codec"); + +static struct hda_codec_preset_list si3054_list = { + .preset = snd_hda_preset_si3054, + .owner = THIS_MODULE, +}; + +static int __init patch_si3054_init(void) +{ + return snd_hda_add_codec_preset(&si3054_list); +} + +static void __exit patch_si3054_exit(void) +{ + snd_hda_delete_codec_preset(&si3054_list); +} + +module_init(patch_si3054_init) +module_exit(patch_si3054_exit) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 70181d5..9e07f44 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -33,7 +33,6 @@ #include <sound/jack.h> #include "hda_codec.h" #include "hda_local.h" -#include "hda_patch.h" #include "hda_beep.h" enum { @@ -5455,7 +5454,7 @@ static int patch_stac9872(struct hda_codec *codec) /* * patch entries */ -struct hda_codec_preset snd_hda_preset_sigmatel[] = { +static struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x83847690, .name = "STAC9200", .patch = patch_stac9200 }, { .id = 0x83847882, .name = "STAC9220 A1", .patch = patch_stac922x }, { .id = 0x83847680, .name = "STAC9221 A1", .patch = patch_stac922x }, @@ -5519,3 +5518,27 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x111d76b7, .name = "92HD71B5X", .patch = patch_stac92hd71bxx }, {} /* terminator */ }; + +MODULE_ALIAS("snd-hda-codec-id:8384*"); +MODULE_ALIAS("snd-hda-codec-id:111d*"); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("IDT/Sigmatel HD-audio codec"); + +static struct hda_codec_preset_list sigmatel_list = { + .preset = snd_hda_preset_sigmatel, + .owner = THIS_MODULE, +}; + +static int __init patch_sigmatel_init(void) +{ + return snd_hda_add_codec_preset(&sigmatel_list); +} + +static void __exit patch_sigmatel_exit(void) +{ + snd_hda_delete_codec_preset(&sigmatel_list); +} + +module_init(patch_sigmatel_init) +module_exit(patch_sigmatel_exit) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 05182be..6e4d01d 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -47,7 +47,6 @@ #include <sound/asoundef.h> #include "hda_codec.h" #include "hda_local.h" -#include "hda_patch.h" /* amp values */ #define AMP_VAL_IDX_SHIFT 19 @@ -3249,7 +3248,7 @@ static int patch_vt1702(struct hda_codec *codec) /* * patch entries */ -struct hda_codec_preset snd_hda_preset_via[] = { +static struct hda_codec_preset snd_hda_preset_via[] = { { .id = 0x11061708, .name = "VIA VT1708", .patch = patch_vt1708}, { .id = 0x11061709, .name = "VIA VT1708", .patch = patch_vt1708}, { .id = 0x1106170A, .name = "VIA VT1708", .patch = patch_vt1708}, @@ -3320,3 +3319,26 @@ struct hda_codec_preset snd_hda_preset_via[] = { .patch = patch_vt1702}, {} /* terminator */ }; + +MODULE_ALIAS("snd-hda-codec-id:1106*"); + +static struct hda_codec_preset_list via_list = { + .preset = snd_hda_preset_via, + .owner = THIS_MODULE, +}; + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("VIA HD-audio codec"); + +static int __init patch_via_init(void) +{ + return snd_hda_add_codec_preset(&via_list); +} + +static void __exit patch_via_exit(void) +{ + snd_hda_delete_codec_preset(&via_list); +} + +module_init(patch_via_init) +module_exit(patch_via_exit) -- 1.6.0.4

3 7

[alsa-devel] [PATCH] ALSA: hda - Don't touch non-existent port f on 4-port 92hd71bxx codecs
by Herton Ronaldo Krzesinski 09 Feb '09

09 Feb '09

When checking for input amps on pins 0x0a, 0x0d and 0x0f, and initializing them for 92hd71xxx codec models, we must skip nid 0x0f for 4-port models too like with 5-port models, as it is unused (nid 0x0f is vendor reserved in 4-port models). Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d16d5c6..2f4e090 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5072,6 +5072,8 @@ again: switch (codec->vendor_id) { case 0x111d76b6: /* 4 Port without Analog Mixer */ case 0x111d76b7: + unmute_init++; + /* fallthru */ case 0x111d76b4: /* 6 Port without Analog Mixer */ case 0x111d76b5: memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_nomixer, -- 1.6.1.2

2 1

[alsa-devel] [PATCH] ASoC: count reaches 10001, not 10000.
by Roel Kluin 08 Feb '09

08 Feb '09

With a postfix increment count reaches 10001, not 10000. diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c index 7f2a5e1..edfbdc0 100644 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -114,7 +114,7 @@ static int snd_ad73311_configure(void) SSYNC(); /* When TUVF is set, the data is already send out */ - while (!(status & TUVF) && count++ < 10000) { + while (!(status & TUVF) && ++count < 10000) { udelay(1); status = bfin_read_SPORT_STAT(); SSYNC(); @@ -123,7 +123,7 @@ static int snd_ad73311_configure(void) SSYNC(); local_irq_enable(); - if (count == 10000) { + if (count >= 10000) { printk(KERN_ERR "ad73311: failed to configure codec\n"); return -1; }

3 2

[alsa-devel] [PATCH] Add initial support of Mitac mioa701 device SoC.
by Robert Jarzmik 08 Feb '09

08 Feb '09

This machine driver enables sound functions on Mitac mio a701 smartphone. Build upon ASoC v1, it handles : - rear speaker - front speaker - microphone - GSM A global "Mio Mode" switch is not yet provided to cope with audio path setup. As balance on audio chip line is no more assured, an incorrect setup can produce a lot of heat and even fry the battery behind the wm9713 and the speaker amplificator. It doesn't cope with : - headset jack - mio master mode - master volume control This driver is backported from ASoc v2, and amputated from scenario setups and master volume control. Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- sound/soc/pxa/Kconfig | 9 ++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/mioa701_wm9713.c | 250 ++++++++++++++++++++++++++++++++++++++++ 3 files changed, 261 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/mioa701_wm9713.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index f82e106..b385404 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -97,3 +97,12 @@ config SND_SOC_ZYLONITE help Say Y if you want to add support for SoC audio on the Marvell Zylonite reference platform. + +config SND_PXA2XX_SOC_MIOA701 + tristate "SoC Audio support for MIO A701" + depends on SND_PXA2XX_SOC && MACH_MIOA701 + select SND_PXA2XX_SOC_AC97 + select SND_SOC_WM9713 + help + Say Y if you want to add support for SoC audio on the + MIO A701. diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 08a9f27..9a6d27c 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -18,6 +18,7 @@ snd-soc-spitz-objs := spitz.o snd-soc-em-x270-objs := em-x270.o snd-soc-palm27x-objs := palm27x.o snd-soc-zylonite-objs := zylonite.o +snd-soc-mioa701-objs := mioa701_wm9713.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o @@ -26,4 +27,5 @@ obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o +obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c new file mode 100644 index 0000000..839f928 --- /dev/null +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -0,0 +1,250 @@ +/* + * Handles the Mitac mioa701 SoC system + * + * Copyright (C) 2008 Robert Jarzmik + * + * 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 in version 2 of the License. + * + * 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 + * + * This is a little schema of the sound interconnections : + * + * Sagem X200 Wolfson WM9713 + * +--------+ +-------------------+ Rear Speaker + * | | | | /-+ + * | +--->----->---+MONOIN SPKL+--->----+-+ | + * | GSM | | | | | | + * | +--->----->---+PCBEEP SPKR+--->----+-+ | + * | CHIP | | | \-+ + * | +---<-----<---+MONO | + * | | | | Front Speaker + * +--------+ | | /-+ + * | HPL+--->----+-+ | + * | | | | | + * | OUT3+--->----+-+ | + * | | \-+ + * | | + * | | Front Micro + * | | + + * | MIC1+-----<--+o+ + * | | + + * +-------------------+ --- + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> + +#include <asm/mach-types.h> +#include <mach/audio.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/ac97_codec.h> + +#include "pxa2xx-pcm.h" +#include "pxa2xx-ac97.h" +#include "../codecs/wm9713.h" + +#define ARRAY_AND_SIZE(x) (x), ARRAY_SIZE(x) + +#define AC97_GPIO_PULL 0x58 + +/* Use GPIO8 for rear speaker amplificator */ +static int rear_amp_power(struct snd_soc_codec *codec, int power) +{ + unsigned short reg; + + if (power) { + reg = snd_soc_read(codec, AC97_GPIO_CFG); + snd_soc_write(codec, AC97_GPIO_CFG, reg | 0x0100); + reg = snd_soc_read(codec, AC97_GPIO_PULL); + snd_soc_write(codec, AC97_GPIO_PULL, reg | (1<<15)); + } else { + reg = snd_soc_read(codec, AC97_GPIO_CFG); + snd_soc_write(codec, AC97_GPIO_CFG, reg & ~0x0100); + reg = snd_soc_read(codec, AC97_GPIO_PULL); + snd_soc_write(codec, AC97_GPIO_PULL, reg & ~(1<<15)); + } + + return 0; +} + +static int rear_amp_event(struct snd_soc_dapm_widget *widget, + struct snd_kcontrol *kctl, int event) +{ + struct snd_soc_codec *codec = widget->codec; + + return rear_amp_power(codec, SND_SOC_DAPM_EVENT_ON(event)); +} + +/* mioa701 machine dapm widgets */ +static const struct snd_soc_dapm_widget mioa701_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Front Speaker", NULL), + SND_SOC_DAPM_SPK("Rear Speaker", rear_amp_event), + SND_SOC_DAPM_MIC("Headset", NULL), + SND_SOC_DAPM_LINE("GSM Line Out", NULL), + SND_SOC_DAPM_LINE("GSM Line In", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Front Mic", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Call Mic */ + {"Mic Bias", NULL, "Front Mic"}, + {"MIC1", NULL, "Mic Bias"}, + + /* Headset Mic */ + {"LINEL", NULL, "Headset Mic"}, + {"LINER", NULL, "Headset Mic"}, + + /* GSM Module */ + {"MONOIN", NULL, "GSM Line Out"}, + {"PCBEEP", NULL, "GSM Line Out"}, + {"GSM Line In", NULL, "MONO"}, + + /* headphone connected to HPL, HPR */ + {"Headset", NULL, "HPL"}, + {"Headset", NULL, "HPR"}, + + /* front speaker connected to HPL, OUT3 */ + {"Front Speaker", NULL, "HPL"}, + {"Front Speaker", NULL, "OUT3"}, + + /* rear speaker connected to SPKL, SPKR */ + {"Rear Speaker", NULL, "SPKL"}, + {"Rear Speaker", NULL, "SPKR"}, +}; + +static int mioa701_wm9713_init(struct snd_soc_codec *codec) +{ + unsigned short reg; + + /* Add mioa701 specific widgets */ + snd_soc_dapm_new_controls(codec, ARRAY_AND_SIZE(mioa701_dapm_widgets)); + + /* Set up mioa701 specific audio path audio_mapnects */ + snd_soc_dapm_add_routes(codec, ARRAY_AND_SIZE(audio_map)); + + /* Prepare GPIO8 for rear speaker amplificator */ + reg = codec->read(codec, AC97_GPIO_CFG); + codec->write(codec, AC97_GPIO_CFG, reg | 0x0100); + + /* Prepare MIC input */ + reg = codec->read(codec, AC97_3D_CONTROL); + codec->write(codec, AC97_3D_CONTROL, reg | 0xc000); + + snd_soc_dapm_enable_pin(codec, "Front Speaker"); + snd_soc_dapm_enable_pin(codec, "Rear Speaker"); + snd_soc_dapm_enable_pin(codec, "Front Mic"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_ops mioa701_ops; + +static struct snd_soc_dai_link mioa701_dai[] = { + { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], + .codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI], + .init = mioa701_wm9713_init, + .ops = &mioa701_ops, + }, + { + .name = "AC97 Aux", + .stream_name = "AC97 Aux", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], + .codec_dai = &wm9713_dai[WM9713_DAI_AC97_AUX], + .ops = &mioa701_ops, + }, +}; + +static struct snd_soc_card mioa701 = { + .name = "MioA701", + .platform = &pxa2xx_soc_platform, + .dai_link = mioa701_dai, + .num_links = ARRAY_SIZE(mioa701_dai), +}; + +static struct snd_soc_device mioa701_snd_devdata = { + .card = &mioa701, + .codec_dev = &soc_codec_dev_wm9713, +}; + +static struct platform_device *mioa701_snd_device; + +static int mioa701_wm9713_probe(struct platform_device *pdev) +{ + int ret; + + if (!machine_is_mioa701()) + return -ENODEV; + + dev_warn(&pdev->dev, "Be warned that incorrect mixes/muxes setup will" + "lead to overheating and possible destruction of your device." + "Do not use without a good knowledge of mio's board design!\n"); + + mioa701_snd_device = platform_device_alloc("soc-audio", -1); + if (!mioa701_snd_device) + return -ENOMEM; + + platform_set_drvdata(mioa701_snd_device, &mioa701_snd_devdata); + mioa701_snd_devdata.dev = &mioa701_snd_device->dev; + + ret = platform_device_add(mioa701_snd_device); + if (!ret) + return 0; + + platform_device_put(mioa701_snd_device); + return ret; +} + +static int __devexit mioa701_wm9713_remove(struct platform_device *pdev) +{ + platform_device_unregister(mioa701_snd_device); + return 0; +} + +static struct platform_driver mioa701_wm9713_driver = { + .probe = mioa701_wm9713_probe, + .remove = __devexit_p(mioa701_wm9713_remove), + .driver = { + .name = "mioa701-wm9713", + .owner = THIS_MODULE, + }, +}; + +static int __init mioa701_asoc_init(void) +{ + return platform_driver_register(&mioa701_wm9713_driver); +} + +static void __exit mioa701_asoc_exit(void) +{ + platform_driver_unregister(&mioa701_wm9713_driver); +} + +module_init(mioa701_asoc_init); +module_exit(mioa701_asoc_exit); + +/* Module information */ +MODULE_AUTHOR("Robert Jarzmik (rjarzmik(a)free.fr)"); +MODULE_DESCRIPTION("ALSA SoC WM9713 MIO A701"); +MODULE_LICENSE("GPL"); -- 1.5.6.5

2 1

[alsa-devel] Thank for your efforts.
by Guy Stalnaker 08 Feb '09

08 Feb '09

Takashi, Herton, Tim, Jeanne-Pierre, etc. My thanks for your efforts to get sound working on the internal speakers on HP Pavilion Dv4 laptops. I have a brand new1225 (one week old) and Ubuntu 8.10 works splendidly in all the things I need (wireless, graphics, etc.). Sound works, however, only on the front headphone jack, but not through the internal speakers. I offer myself as another test platform for your test code. To date I've installed the lastest snapshot, have model=hp-dv5 in /etc/modprobe.d/alsa-base, and I've tried Takashi's hda-verb option to modify the pin. That does change the pin value, but it has no affect, internal speaker's remain dead. Best regards, Guy -- Guy Stalnaker, I^2@DOIT, 1210 West Dayton Street, Room 3209 CSS, Madison WI 53719-1220, jstalnak(a)wisc.edu, work 608.263.8035, cell 608.235.4718, fax 608.265.6681, page page-guy(a)watchdog.doit.wisc.edu

1 0

[alsa-devel] RME HDSP rev 37 does not work on 64-bit linux
by Dmitry Baikov 08 Feb '09

08 Feb '09

Hello, recently I upgraded firmware for my RME HDSP (Cardbus + Multiface II) to recent version, to be able to use it under WinXP 64, since there's no 64-bit windows driver for old hdsp firmware. And the results under linux are surprising: My main OS is Ubuntu 8.10 64-bit, and there hdspmixer fails with the message: Card 1 : RME Hammerfall DSP at 0x130000000, irq 18 Uninitialized HDSP card found. Use hdsploader to upload configuration data to the card. No Hammerfall DSP card found. Trying to run hdsploader gives: Card 1 : RME Hammerfall DSP at 0x130000000, irq 18 Upload firmware for card hw:1 Hwdep ioctl error on card hw:1 : Input/output error. dmesg: Hammerfall-DSP: wait for FIFO status <= 0 failed after 30 iterations This happens even if I boot into winxp first to initialize multiface, then reboot into linux and do not run hdsploader. BUT: under 32-bit ubuntu 8.10 with the same kernel 2.6.27-11-generic and the same version of alsa-*: 1.0.17-0ubuntu1, everything works fine both with cold boot + hdsploader and winxp -> reboot-> no hdsploader. Regards, Dmitry.

4 8

[alsa-devel] redirect stream of audio
by Hartman Mario Andres 07 Feb '09

07 Feb '09

1 0

[PATCH] hda: Add mising beep controls for realtek alc268 codec.
by Kusanagi Kouichi 07 Feb '09

07 Feb '09

So far, some models didn't add a beep control since they don't use analog beep in. Signed-off-by: Kusanagi Kouichi <slash(a)ma.neweb.ne.jp> --- sound/pci/hda/patch_realtek.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f594a09..2fa26f9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11887,7 +11887,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { static struct alc_config_preset alc268_presets[] = { [ALC267_QUANTA_IL1] = { - .mixers = { alc267_quanta_il1_mixer }, + .mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer }, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc267_quanta_il1_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -11969,7 +11969,7 @@ static struct alc_config_preset alc268_presets[] = { }, [ALC268_ACER_ASPIRE_ONE] = { .mixers = { alc268_acer_aspire_one_mixer, - alc268_capture_alt_mixer }, + alc268_capture_alt_mixer, alc268_beep_mixer }, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_acer_aspire_one_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), -- 1.5.6.5 --UlVJffcvxoiEqYs2--

1 0

[alsa-devel] Internal Mic of Philips SPC 1300NC Webcam not recording (same with SPC 1000 and 620)
by Sinisa Dukanovic 07 Feb '09

07 Feb '09

Hi everyone, i can't record sound from the internal mic of my Phillips SPC1300 Webcam. I have already filed a bug report at: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3927 where i posted a lot of information. In addition to that, you can find the latest results of alsa-info.sh here: http://www.alsa-project.org/db/?f=a3fabdb13269a50e5068a76b713a0d45f1794aaf Is there anything (besides those infos in the bugreport) that i can do to help fixing this bug? regards, SD

1 0

[alsa-devel] [PATCH 2.6.30] ASoC: optimize init sequence of Freescale MPC8610 sound drivers
by Timur Tabi 06 Feb '09

06 Feb '09

In the Freescale MPC8610 sound drivers, relocate all code from the _prepare functions into the corresponding _hw_params functions. These drivers assumed that the sample size is known in the _prepare function and not in the _hw_params function, but this is not true. Move the code in fsl_dma_prepare() into fsl_dma_hw_param(). Create fsl_ssi_hw_params() and move the code from fsl_ssi_prepare() into it. Turn off snooping for DMA operations to/from I/O registers, since that's not necessary. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/fsl_dma.c | 178 ++++++++++++++++++++++------------------------ sound/soc/fsl/fsl_ssi.c | 19 +++--- 2 files changed, 94 insertions(+), 103 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 64993ed..58a3fa4 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -464,11 +464,7 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) sizeof(struct fsl_dma_link_descriptor); for (i = 0; i < NUM_DMA_LINKS; i++) { - struct fsl_dma_link_descriptor *link = &dma_private->link[i]; - - link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); - link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); - link->next = cpu_to_be64(temp_link); + dma_private->link[i].next = cpu_to_be64(temp_link); temp_link += sizeof(struct fsl_dma_link_descriptor); } @@ -525,79 +521,9 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) * This function obtains hardware parameters about the opened stream and * programs the DMA controller accordingly. * - * Note that due to a quirk of the SSI's STX register, the target address - * for the DMA operations depends on the sample size. So we don't program - * the dest_addr (for playback -- source_addr for capture) fields in the - * link descriptors here. We do that in fsl_dma_prepare() - */ -static int fsl_dma_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct fsl_dma_private *dma_private = runtime->private_data; - - dma_addr_t temp_addr; /* Pointer to next period */ - - unsigned int i; - - /* Get all the parameters we need */ - size_t buffer_size = params_buffer_bytes(hw_params); - size_t period_size = params_period_bytes(hw_params); - - /* Initialize our DMA tracking variables */ - dma_private->period_size = period_size; - dma_private->num_periods = params_periods(hw_params); - dma_private->dma_buf_end = dma_private->dma_buf_phys + buffer_size; - dma_private->dma_buf_next = dma_private->dma_buf_phys + - (NUM_DMA_LINKS * period_size); - if (dma_private->dma_buf_next >= dma_private->dma_buf_end) - dma_private->dma_buf_next = dma_private->dma_buf_phys; - - /* - * The actual address in STX0 (destination for playback, source for - * capture) is based on the sample size, but we don't know the sample - * size in this function, so we'll have to adjust that later. See - * comments in fsl_dma_prepare(). - * - * The DMA controller does not have a cache, so the CPU does not - * need to tell it to flush its cache. However, the DMA - * controller does need to tell the CPU to flush its cache. - * That's what the SNOOP bit does. - * - * Also, even though the DMA controller supports 36-bit addressing, for - * simplicity we currently support only 32-bit addresses for the audio - * buffer itself. - */ - temp_addr = substream->dma_buffer.addr; - - for (i = 0; i < NUM_DMA_LINKS; i++) { - struct fsl_dma_link_descriptor *link = &dma_private->link[i]; - - link->count = cpu_to_be32(period_size); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - link->source_addr = cpu_to_be32(temp_addr); - else - link->dest_addr = cpu_to_be32(temp_addr); - - temp_addr += period_size; - } - - return 0; -} - -/** - * fsl_dma_prepare - prepare the DMA registers for playback. - * - * This function is called after the specifics of the audio data are known, - * i.e. snd_pcm_runtime is initialized. - * - * In this function, we finish programming the registers of the DMA - * controller that are dependent on the sample size. - * - * One of the drawbacks with big-endian is that when copying integers of - * different sizes to a fixed-sized register, the address to which the - * integer must be copied is dependent on the size of the integer. + * One drawback of big-endian is that when copying integers of different + * sizes to a fixed-sized register, the address to which the integer must be + * copied is dependent on the size of the integer. * * For example, if P is the address of a 32-bit register, and X is a 32-bit * integer, then X should be copied to address P. However, if X is a 16-bit @@ -613,22 +539,58 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, * and 8 bytes at a time). So we do not support packed 24-bit samples. * 24-bit data must be padded to 32 bits. */ -static int fsl_dma_prepare(struct snd_pcm_substream *substream) +static int fsl_dma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) { struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_dma_private *dma_private = runtime->private_data; + + /* Number of bits per sample */ + unsigned int sample_size = + snd_pcm_format_physical_width(params_format(hw_params)); + + /* Number of bytes per frame */ + unsigned int frame_size = 2 * (sample_size / 8); + + /* Bus address of SSI STX register */ + dma_addr_t ssi_sxx_phys = dma_private->ssi_sxx_phys; + + /* Size of the DMA buffer, in bytes */ + size_t buffer_size = params_buffer_bytes(hw_params); + + /* Number of bytes per period */ + size_t period_size = params_period_bytes(hw_params); + + /* Pointer to next period */ + dma_addr_t temp_addr = substream->dma_buffer.addr; + + /* Pointer to DMA controller */ struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel; - u32 mr; + + u32 mr; /* DMA Mode Register */ + unsigned int i; - dma_addr_t ssi_sxx_phys; /* Bus address of SSI STX register */ - unsigned int frame_size; /* Number of bytes per frame */ - ssi_sxx_phys = dma_private->ssi_sxx_phys; + /* Initialize our DMA tracking variables */ + dma_private->period_size = period_size; + dma_private->num_periods = params_periods(hw_params); + dma_private->dma_buf_end = dma_private->dma_buf_phys + buffer_size; + dma_private->dma_buf_next = dma_private->dma_buf_phys + + (NUM_DMA_LINKS * period_size); + + if (dma_private->dma_buf_next >= dma_private->dma_buf_end) + /* This happens if the number of periods == NUM_DMA_LINKS */ + dma_private->dma_buf_next = dma_private->dma_buf_phys; mr = in_be32(&dma_channel->mr) & ~(CCSR_DMA_MR_BWC_MASK | CCSR_DMA_MR_SAHTS_MASK | CCSR_DMA_MR_DAHTS_MASK); - switch (runtime->sample_bits) { + /* Due to a quirk of the SSI's STX register, the target address + * for the DMA operations depends on the sample size. So we calculate + * that offset here. While we're at it, also tell the DMA controller + * how much data to transfer per sample. + */ + switch (sample_size) { case 8: mr |= CCSR_DMA_MR_DAHTS_1 | CCSR_DMA_MR_SAHTS_1; ssi_sxx_phys += 3; @@ -641,12 +603,12 @@ static int fsl_dma_prepare(struct snd_pcm_substream *substream) mr |= CCSR_DMA_MR_DAHTS_4 | CCSR_DMA_MR_SAHTS_4; break; default: + /* We should never get here */ dev_err(substream->pcm->card->dev, - "unsupported sample size %u\n", runtime->sample_bits); + "unsupported sample size %u\n", sample_size); return -EINVAL; } - frame_size = runtime->frame_bits / 8; /* * BWC should always be a multiple of the frame size. BWC determines * how many bytes are sent/received before the DMA controller checks the @@ -655,7 +617,6 @@ static int fsl_dma_prepare(struct snd_pcm_substream *substream) * capture, the receive FIFO is triggered when it contains one frame, so * we want to receive one frame at a time. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) mr |= CCSR_DMA_MR_BWC(2 * frame_size); else @@ -663,16 +624,48 @@ static int fsl_dma_prepare(struct snd_pcm_substream *substream) out_be32(&dma_channel->mr, mr); - /* - * Program the address of the DMA transfer to/from the SSI. - */ for (i = 0; i < NUM_DMA_LINKS; i++) { struct fsl_dma_link_descriptor *link = &dma_private->link[i]; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + link->count = cpu_to_be32(period_size); + + /* Even though the DMA controller supports 36-bit addressing, + * for simplicity we allow only 32-bit addresses for the audio + * buffer itself. This was enforced in fsl_dma_new() with the + * DMA mask. + * + * The snoop bit tells the DMA controller whether it should tell + * the ECM to snoop during a read or write to an address. For + * audio, we use DMA to transfer data between memory and an I/O + * device (the SSI's STX0 or SRX0 register). Snooping is only + * needed if there is a cache, so we need to snoop memory + * addresses only. For playback, that means we snoop the source + * but not the destination. For capture, we snoop the + * destination but not the source. + * + * Note that failing to snoop properly is unlikely to cause + * cache incoherency if the period size is larger than the + * size of L1 cache. This is because filling in one period will + * flush out the data for the previous period. So if you + * increased period_bytes_min to a large enough size, you might + * get more performance by not snooping, and you'll still be + * okay. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + link->source_addr = cpu_to_be32(temp_addr); + link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); + link->dest_addr = cpu_to_be32(ssi_sxx_phys); - else + link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_NOSNOOP); + } else { link->source_addr = cpu_to_be32(ssi_sxx_phys); + link->source_attr = cpu_to_be32(CCSR_DMA_ATR_NOSNOOP); + + link->dest_addr = cpu_to_be32(temp_addr); + link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); + } + + temp_addr += period_size; } return 0; @@ -808,7 +801,6 @@ static struct snd_pcm_ops fsl_dma_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = fsl_dma_hw_params, .hw_free = fsl_dma_hw_free, - .prepare = fsl_dma_prepare, .pointer = fsl_dma_pointer, }; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c6d6eb7..6844009 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -400,7 +400,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, } /** - * fsl_ssi_prepare: prepare the SSI. + * fsl_ssi_hw_params - program the sample size * * Most of the SSI registers have been programmed in the startup function, * but the word length must be programmed here. Unfortunately, programming @@ -412,20 +412,19 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * Note: The SxCCR.DC and SxCCR.PM bits are only used if the SSI is the * clock master. */ -static int fsl_ssi_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *cpu_dai) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; - - struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + struct fsl_ssi_private *ssi_private = cpu_dai->private_data; if (substream == ssi_private->first_stream) { + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + unsigned int sample_size = + snd_pcm_format_width(params_format(hw_params)); u32 wl; /* The SSI should always be disabled at this points (SSIEN=0) */ - wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format)); + wl = CCSR_SSI_SxCCR_WL(sample_size); /* In synchronous mode, the SSI uses STCCR for capture */ clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); @@ -579,7 +578,7 @@ static struct snd_soc_dai fsl_ssi_dai_template = { }, .ops = { .startup = fsl_ssi_startup, - .prepare = fsl_ssi_prepare, + .hw_params = fsl_ssi_hw_params, .shutdown = fsl_ssi_shutdown, .trigger = fsl_ssi_trigger, .set_sysclk = fsl_ssi_set_sysclk, -- 1.5.5

2 1

[alsa-devel] [2.6.29-rc3][PATCH 1/2] ASoC: TLV320AIC3X: Fix kcontrol's private value use in put callback
by Jarkko Nikula 06 Feb '09

06 Feb '09

From: Eero Nurkkala <ext-eero.nurkkala(a)nokia.com> Function snd_soc_dapm_put_volsw_aic3x misuses the kcontrol's private value by still accessing it as bitfields even SOC_SINGLE_VALUE constructs it as a pointer into struct soc_mixer_control after the commit 4eaa9819dc08d7bfd1065ce530e31b18a119dcaf. This was causing arbitrary register writes when touching the controls defined with SOC_DAPM_SINGLE_AIC3X. Signed-off-by: Eero Nurkkala <ext-eero.nurkkala(a)nokia.com> Acked-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- sound/soc/codecs/tlv320aic3x.c | 11 +++++++---- 1 files changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index b47a749..aea0cb7 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -165,10 +165,13 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0x0f; - int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0x01; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + int max = mc->max; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; unsigned short val, val_mask; int ret; struct snd_soc_dapm_path *path; -- 1.5.6.5

2 2

Re: [alsa-devel] S/PDIF-Out on Asus M3A-H/HDMI with ALC1200]
by Thomas Schneider 06 Feb '09

06 Feb '09

Hi, here is the output from alsa-info.sh http://www.alsa-project.org/db/?f=8282910dc807e1f141896886cd5f8fd574117000 I found another small mistake. In the Mixer my Soundcard is shown as "Realtek ALC888". When i switch this to lines in patch_realtek.c then it is shown as "Realtek ALC1200". static struct hda_codec_preset snd_hda_preset_realtek[] = { ...... { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc883 }, { .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200", .patch = patch_alc883 }, ..... Thomas Am 25.12.2008 05:31, schrieb Wu Fengguang: > Hi Thomas, > > Could you provide more information by running > http://www.alsa-project.org/alsa-info.sh > > I'm particularity interested in the full dmesg and contents of > /proc/asound/card?/*\#? > > Thanks, > Fengguang > > On Tue, Dec 23, 2008 at 12:22:29PM +0100, Thomas Schneider wrote: > >> Hi, >> >> i am looking for a way to make Digital-Audio work with the external >> and internal S/PDIF-Output of my Motherboard Asus M3A-H/HDMI with >> ALC1200 Soundard. I use the latest unstable snapshot and i can´t >> make it work at the same time. >> >> On the external S/PDIF-Output is my Digital Receiver connected, >> on the internal one my GeForce 8500GT to get sound over HDMI >> on my LCT-TV. >> >> When i load the driver without any model-options my Soundcard is >> detected as Asus P5Q-EM HDMI >> >> ALSA /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:2716: >> hda_codec: model 'asus-p5q' is selected for config 1043:82fe (Asus >> P5Q-EM HDMI) >> [ 6.721700] ALSA >> /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot >> find slave Headphone Playback Volume, skipped >> [ 6.721714] ALSA >> /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot >> find slave Speaker Playback Volume, skipped >> [ 6.721716] ALSA >> /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot >> find slave Mono Playback Volume, skipped >> [ 6.721719] ALSA >> /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot >> find slave Line-Out Playback Volume, skipped >> [ 6.721721] ALSA >> /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot >> find slave PCM Playback Volume, skipped >> [ 6.721729] ALSA >> /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot >> find slave Speaker Playback Switch, skipped >> [ 6.721731] ALSA >> /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot >> find slave Mono Playback Switch, skipped >> >> Now i have sound on my LCD-TV over HDMI (internal S/PDIF-Out) >> >> When i load the driver with model=6stack-dig i have sound on my >> Digital-Receiver (external S/PDIF-Out). >> >> In the mailinglist-archive or in the doku i have seen, that the only >> difference >> between this models is the DIGOUT_NID >> >> asus-p5q = 0x10 >> 6stack-dig = 0x06 >> >> I there a way to get both output to work at the same time so i have >> sound on my LCD-TV and my Digital-Receiver ? >> >> >> Regards >> >> Thomas >> _______________________________________________ >> Alsa-devel mailing list >> Alsa-devel(a)alsa-project.org >> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >> > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >

3 12

[alsa-devel] No sound for HP Pavilion DV4z
by Andrew Eleneski 06 Feb '09

06 Feb '09

Hi, I am running Kubuntu 8.10 64-bit and have no sound. Using the distribution supplied Alsa version, I was at one point able to have sound through just the headphone jacks, but after using this script ( http://ubuntuforums.org/showthread.php?t=1046137&highlight=alsa+upgrade) and trying to upgrade to the latest version 1.0.19, I have no sound at all. Attached is my alsa-info.txt file to help in debugging what is wrong. 1. Name your Ubuntu revision: Kubuntu 8.10 64-bit 2. Kernel revision: 2.6.27-11-generic 3. Alsa revision: 1.0.19 4. A bit of background what you've done resp. done before: I have tried multitudes of values in /etc/modprobe.d/alsa-base, nothing seems to work. Saw in changelog for Alsa 1.0.19 there were many changes geared toward HP dvX series laptops, seems to have no effect with mine. Thanks.

3 3

[alsa-devel] [PATCH v2 0/3] Atmel ALSA drivers
by Hans-Christian Egtvedt 05 Feb '09

05 Feb '09

This patch series is the second revision after the initial review on alsa-devel mailing list. I have addressed the feedback and fixed up the drivers appropriatly, at least I hope so. I've settled to put the drivers in the more common sound/atmel and added an "Atmel devices (AVR32 and AT91)" menu entry in the ALSA part of kconfig. No drivers are available for AT91 yet, but the AC97C driver should be quite simple to add support for AT91 devices as well. The patches are formatted against v2.6.29-rc3 tag from Linux repository. In addition a vital DMA patch[1] is needed, which adds a cyclic DMA interface to the DesignWare DMA controller. 1: http://avr32linux.org/archives/kernel/2009-February/002088.html http://avr32linux.org/archives/kernel/2009-February/002087.html -- Best regards, Hans-Christian Egtvedt

2 6

[alsa-devel] [PATCH] alsa: snd_pcm_new api cleanup
by Tim Blechmann 05 Feb '09

05 Feb '09

Impact: cleanup snd_pcm_new takes a char *id argument, although it is not modifying the string. it can therefore be declared as const char *id. Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- sound/core/pcm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 192a433..583453e 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -692,7 +692,7 @@ EXPORT_SYMBOL(snd_pcm_new_stream); * * Returns zero if successful, or a negative error code on failure. */ -int snd_pcm_new(struct snd_card *card, char *id, int device, +int snd_pcm_new(struct snd_card *card, const char *id, int device, int playback_count, int capture_count, struct snd_pcm ** rpcm) { -- 1.5.6.3

2 4

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 05 Feb '09

05 Feb '09

The following changes since commit 395707212a53999427bd160abd6541bffe062033: Takashi Iwai (1): Merge branch 'fix/asoc' into topic/asoc are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Philipp Zabel (2): ASoC: uda1380: split set_dai_fmt into _both, _playback and _capture variants ASoC: pxa-ssp: fix SSP port request Vasily Khoruzhick (1): ASoC: Don't unconditionally use the PLL in UDA1380 sound/soc/codecs/uda1380.c | 72 ++++++++++++++++++++++++++++++++++++++----- sound/soc/pxa/pxa-ssp.c | 12 ++++--- 2 files changed, 70 insertions(+), 14 deletions(-)

2 1

[alsa-devel] Tascam US-1641 Firmware File
by Donnie Menig 05 Feb '09

05 Feb '09

Sorry all, I forgot to actually attach the file in my previous post. Here it is in this post.

3 2

[alsa-devel] [PATCH] ASoC: pxa-ssp: disable only when the ssp dai is active
by Eric Miao 05 Feb '09

05 Feb '09

2 1

[PATCH] ASoC: pxa-ssp: disable only when the ssp dai is active
by Eric Miao 05 Feb '09

05 Feb '09

Signed-off-by: Eric Miao <eric.miao(a)marvell.com> --- Typo? sound/soc/pxa/pxa-ssp.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 4a973ab..3f12b45 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -238,7 +238,7 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct ssp_priv *priv = cpu_dai->private_data; - if (!cpu_dai->active) { + if (cpu_dai->active) { ssp_disable(&priv->dev); clk_disable(priv->dev.ssp->clk); } -- 1.6.0.4

1 0

[alsa-devel] Write an alsa driver for the Lexicon Omega
by Fernando Moro 05 Feb '09

05 Feb '09

I would be very gratefull if someone could write an alsa driver for the Lexicon lambda soundcard. I think there are a lot of people with the same problem, but I don't have the necessary knowledge. I can provide the drivers for windows. I hope you can help me, and I hope your news about this device. Thank you. Fernando. _________________________________________________________________ Descubre cómo compartir tus fotos con Windows Live. ¡Pruébalo ya! http://home.live.com/

2 1

[alsa-devel] driver or plugin?
by Hartman Mario Andres 05 Feb '09

05 Feb '09

hi everybody, this is my first post in the list. currently i work in a new project, and i need is some help to finish this. I wrote an application whose take data from serial port or from bluetooth device(spp) and store the data in a raw file(the data is audio). Now i need to do this: the data received from the port, send to internet through voip software like ekiga, kphone, etc. I read writing-an-alsa-driver.pdf, alsa-driver-api.pdf but i dont know how create a source of data, for softwares of voip take data like a microphone and also receive data from these programs. I've been looking serial-u16550.c would be something similar to what the bluez bluetooth headphones Sorry for my english is not very good. greetings

2 1

[alsa-devel] Onkyo SE-200PCI : What can the first pair of RCA output do?
by tycordinal 05 Feb '09

05 Feb '09

Hi, I am running on Fedora 9 / Alsa 1.0.18a. I have tried to play 24bits 6ch PCM file on Onkyo SE-200PCI and the 6ch output from VGA-like connector are very good. Thanks for the good work! I would like to know what is the intention for the first pair of RCA output? Apparently, I am able to get the first two channels out of the 6 channels audio file when I play through plug:surround51, but the remaining four channels were not able to get heard. Could alsamixer or any asound.rc options can make Onkyo SE-200PCI to produce a downmix of 6ch on the first pair of RCA output? Can this be done through the hardware chips themselves, or any asound.rc options can make this work? Or could it be, the first pair of RCA output is really hardwired, like ice1712/se.c has suggested because of WM8740 chip, to always give an exact copy of the first 2 channels out of the 6 channels playback? Thanks.

2 1

[alsa-devel] [PATCH 1/5] snd-usb-caiaq: Send the correct command when setting controls
by Mark Hills 05 Feb '09

05 Feb '09

Fixes a bug where an incorrect command was sent which had no effect on the device. Signed-off-by: Mark Hills <mark(a)pogo.org.uk> Acked-by: Daniel Mack <daniel(a)caiaq.de> --- usb/caiaq/caiaq-control.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/usb/caiaq/caiaq-control.c b/usb/caiaq/caiaq-control.c index 6ac5489..1f9531d 100644 --- a/usb/caiaq/caiaq-control.c +++ b/usb/caiaq/caiaq-control.c @@ -94,7 +94,7 @@ static int control_put(struct snd_kcontrol *kcontrol, if (pos & CNT_INTVAL) { dev->control_state[pos & ~CNT_INTVAL] = ucontrol->value.integer.value[0]; - snd_usb_caiaq_send_command(dev, EP1_CMD_DIMM_LEDS, + snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO, dev->control_state, sizeof(dev->control_state)); } else { if (ucontrol->value.integer.value[0]) -- 1.6.1.2

2 5

[alsa-devel] PATCH: Maxdata Favorit 100XS support
by Michael Schwingen 05 Feb '09

05 Feb '09

Hi, this is my first ALSA patch, so I am not sure if this is the right place & form to post patches? Anyway, here is a patch to enable audio on the Maxdata Favorit 100XS - a compact PC sold by German company Maxdata (probably manufactured by FIC, Transtec Senyo 600/Belinea o.max XS/Lynx Silent PC MP-I *might* be identical, but I have none of those). The Favorit 100XS uses an Intel 945 chipset with a Realtek HD audio codec connected to exactly 2 jacks: one input, and one output jack. Using the standard ALSA driver, output is silent. This patch enables audio input/output on these machines - it is based in part on the acer code in the same file. Signed-off-by: Michael Schwingen <michael(a)schwingen.org> cu Michael

3 2

[alsa-devel] [PATCH 0/3] Add AVR32 ALSA drivers
by Hans-Christian Egtvedt 05 Feb '09

05 Feb '09

From: Hans-Christian Egtvedt <hcegtvedt(a)atmel.com> This series adds support for the ABDAC and AC97C peripherals on AVR32 devices. They also need the DW DMAC cyclic DMA API patch to work[1]. The patch set is made against kernel v2.6.29-rc3. A review would be greatly appreciated, checkpatch.pl gives no warnings and usage works as expected on hardware. Adding these drivers to the queue for the upstream kernel should wait until the cyclic DMA API for the DW DMAC driver is also in the queue for upstream. 1: http://avr32linux.org/archives/kernel/2009-February/002088.html http://avr32linux.org/archives/kernel/2009-February/002087.html -- Best regards, Hans-Christian Egtvedt

4 23

[alsa-devel] [PATCH] hda: Added stac378x digital slave out struct
by Matthew Ranostay 05 Feb '09

05 Feb '09

Added the ADATOut nid to a slave digital outs struct to allow output via the DigOut pin. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- sound/pci/hda/patch_sigmatel.c | 5 +++++ 1 files changed, 5 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a7df81e..58a10ed 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -404,6 +404,10 @@ static hda_nid_t stac922x_mux_nids[2] = { 0x12, 0x13, }; +static hda_nid_t stac927x_slave_dig_outs[2] = { + 0x1f, 0, +}; + static hda_nid_t stac927x_adc_nids[3] = { 0x07, 0x08, 0x09 }; @@ -5218,6 +5222,7 @@ static int patch_stac927x(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + codec->slave_dig_outs = stac927x_slave_dig_outs; spec->num_pins = ARRAY_SIZE(stac927x_pin_nids); spec->pin_nids = stac927x_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_927X_MODELS, -- 1.5.6.3

2 1

[alsa-devel] Creative X-Fi USB 5.1 Driver
by Mandar Joshi 05 Feb '09

05 Feb '09

>>Does this mean thet the X-Fi,Crystalizer features of the USB X-Fi are implemented purely in software? I think this might be the case. I started SniffUSB under WindowsXP and tried changing the settings using the audio console software provided by Creative. When changing the Speaker selection (2.1/4.1/5.1), data is sent over usb to the device. However, when I try to change the settings for the X-Fi Crystalizer or EAX or CMSS-3D or Graphic Equalizer nothing gets sent to the device but the audio output does change and I am able to hear the corresponding effects. And as mentioned in my previous post, the lsusb output doesn't show any effect descriptors or feature units If this is in fact true that all the xtreme fidelity(X-Fi) stuff is happening in software alone, my purchase has been a total waste!! Does anybody on this list have a USB X-Fi 5.1 that they can use to verify my observations? BTW, the mailing list seems to be creating separate threads for every post I make even though the subject line is the same. What am I missing? I'm using Yahoo Mail. Regards Mandar Joshi

2 1

[alsa-devel] Finding the buffer fill level
by Bankim Bhavsar 05 Feb '09

05 Feb '09

Hi Jaroslav, I'm using following as the reference to find buffer fill-level of the playback channel * WRITE_TO_READ: the current fill level of the playback buffer - buffer_size - snd_pcm_avail_update() http://article.gmane.org/gmane.linux.alsa.devel/54017 Infact I use snd_pcm_hwsync() before calling snd_pcm_avail_update() ( I know I can use snd_pcm_avail() but thats only available 1.0.18 onwards) I use snd_pcm_set_buffer_size_near() to configure the hw params and save the returned value of bufffer_size to find the buffer fill level later on. On some sound cards ( snd_pcm_avail_update() > buffer_size) which shouldn't be the case! This throws off our internal polling calculations. Information about the Linux hosts and sound cards where we've seen this issue. Ubuntu 8.10 x64 host with kernel version 2.6.27-11-generic Alsa lib version 1.0.17a 1) **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog] 2) **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC885 Analog [ALC885 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC885 Digital [ALC885 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 Is this a known issue? Let me know if you need more information. Or there is an alternative approach/workaround? Thanks, Bankim.

1 0

[alsa-devel] [PATCH] pcm_oss: AFMT_S24_LE is set twice in return value
by Roel Kluin 04 Feb '09

04 Feb '09

vi sound/core/oss/pcm_oss.c +640 #define AFMT_S24_LE 0x00008000 #define AFMT_S24_BE 0x00010000 ---------------------->8----------------------8<------------------- AFMT_S24_LE is set twice in return value Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index e178366..0a1798e 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1767,7 +1767,7 @@ static int snd_pcm_oss_get_formats(struct snd_pcm_oss_file *pcm_oss_file) AFMT_S8 | AFMT_U16_LE | AFMT_U16_BE | AFMT_S32_LE | AFMT_S32_BE | - AFMT_S24_LE | AFMT_S24_LE | + AFMT_S24_LE | AFMT_S24_BE | AFMT_S24_PACKED; params = kmalloc(sizeof(*params), GFP_KERNEL); if (!params)

2 1

Re: [alsa-devel] No headphone with an Intel 82801I sound card
by Aurélien Croc 04 Feb '09

04 Feb '09

Is there no news about that problem? If it is just a mixer problem, it may not be so hard to get the correct mixer channels available no? (in particular when each point of the problem work with a specific code). Thank you in advance, Best regards, Aurélien On Monday 02 February 2009, Takashi Iwai wrote: > At Mon, 2 Feb 2009 16:56:07 +0100, > > Aurélien Croc wrote: > > > Hmm, other people reported basically such Dell laptops do work with > > > the headphone but not with the speaker output... > > > > With google i found a lot of guys who have the same problem...I never > > seen someone who doesn't have its internal speaker working.. > > See the recent post by Laurent Pinchart. Which reminds me that the problem still hasn't been fixed :-) Is there anything else you would like me to try ? > Did you try the latest snapshot? > > ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sna >pshot.tar.gz Best regards, Laurent Pinchart _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

1 0

[alsa-devel] [PATCH] sscape: drop redundant fields from soundscape struct (3rd rev)
by Krzysztof Helt 04 Feb '09

04 Feb '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The wss_base is disuised parameter for one function. It is converted to function parameter. The code_type is only set but never read. It is removed. The midi_vol is set only to 0 so it does not work as detection of change in midi volume. It is fixed. The xport variable is alias to the port[dev]. Use the port[dev] directly to increase readability. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The sscape_midi_get() now returns the s->midi_vol value. sound/isa/sscape.c | 44 ++++++++++++++++---------------------------- 1 files changed, 16 insertions(+), 28 deletions(-) diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 681e223..33c1258 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -135,8 +135,6 @@ enum card_type { struct soundscape { spinlock_t lock; unsigned io_base; - unsigned wss_base; - int codec_type; int ic_type; enum card_type type; struct resource *io_res; @@ -726,13 +724,7 @@ static int sscape_midi_get(struct snd_kcontrol *kctl, unsigned long flags; spin_lock_irqsave(&s->lock, flags); - set_host_mode_unsafe(s->io_base); - - if (host_write_ctrl_unsafe(s->io_base, CMD_GET_MIDI_VOL, 100)) { - uctl->value.integer.value[0] = host_read_ctrl_unsafe(s->io_base, 100); - } - - set_midi_mode_unsafe(s->io_base); + uctl->value.integer.value[0] = s->midi_vol; spin_unlock_irqrestore(&s->lock, flags); return 0; } @@ -767,6 +759,7 @@ static int sscape_midi_put(struct snd_kcontrol *kctl, change = (host_write_ctrl_unsafe(s->io_base, CMD_SET_MIDI_VOL, 100) && host_write_ctrl_unsafe(s->io_base, ((unsigned char) uctl->value.integer. value[0]) & 127, 100) && host_write_ctrl_unsafe(s->io_base, CMD_XXX_MIDI_VOL, 100)); + s->midi_vol = (unsigned char) uctl->value.integer.value[0] & 127; __skip_change: /* @@ -809,12 +802,11 @@ static unsigned __devinit get_irq_config(int irq) * Perform certain arcane port-checks to see whether there * is a SoundScape board lurking behind the given ports. */ -static int __devinit detect_sscape(struct soundscape *s) +static int __devinit detect_sscape(struct soundscape *s, long wss_io) { unsigned long flags; unsigned d; int retval = 0; - int codec = s->wss_base; spin_lock_irqsave(&s->lock, flags); @@ -830,13 +822,11 @@ static int __devinit detect_sscape(struct soundscape *s) if ((d & 0x80) != 0) goto _done; - if (d == 0) { - s->codec_type = 1; + if (d == 0) s->ic_type = IC_ODIE; - } else if ((d & 0x60) != 0) { - s->codec_type = 2; + else if ((d & 0x60) != 0) s->ic_type = IC_OPUS; - } else + else goto _done; outb(0xfa, ODIE_ADDR_IO(s->io_base)); @@ -856,10 +846,10 @@ static int __devinit detect_sscape(struct soundscape *s) sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); if (s->type == SSCAPE_VIVO) - codec += 4; + wss_io += 4; /* wait for WSS codec */ for (d = 0; d < 500; d++) { - if ((inb(codec) & 0x80) == 0) + if ((inb(wss_io) & 0x80) == 0) break; spin_unlock_irqrestore(&s->lock, flags); msleep(1); @@ -1057,7 +1047,6 @@ static int __devinit create_sscape(int dev, struct snd_card *card) unsigned dma_cfg; unsigned irq_cfg; unsigned mpu_irq_cfg; - unsigned xport; struct resource *io_res; struct resource *wss_res; unsigned long flags; @@ -1077,15 +1066,15 @@ static int __devinit create_sscape(int dev, struct snd_card *card) printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); return -ENXIO; } - xport = port[dev]; /* * Grab IO ports that we will need to probe so that we * can detect and control this hardware ... */ - io_res = request_region(xport, 8, "SoundScape"); + io_res = request_region(port[dev], 8, "SoundScape"); if (!io_res) { - snd_printk(KERN_ERR "sscape: can't grab port 0x%x\n", xport); + snd_printk(KERN_ERR + "sscape: can't grab port 0x%lx\n", port[dev]); return -EBUSY; } wss_res = NULL; @@ -1112,10 +1101,9 @@ static int __devinit create_sscape(int dev, struct snd_card *card) spin_lock_init(&sscape->fwlock); sscape->io_res = io_res; sscape->wss_res = wss_res; - sscape->io_base = xport; - sscape->wss_base = wss_port[dev]; + sscape->io_base = port[dev]; - if (!detect_sscape(sscape)) { + if (!detect_sscape(sscape, wss_port[dev])) { printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", sscape->io_base); err = -ENODEV; goto _release_dma; @@ -1188,11 +1176,11 @@ static int __devinit create_sscape(int dev, struct snd_card *card) } #define MIDI_DEVNUM 0 if (sscape->type != SSCAPE_VIVO) { - err = create_mpu401(card, MIDI_DEVNUM, xport, mpu_irq[dev]); + err = create_mpu401(card, MIDI_DEVNUM, port[dev], mpu_irq[dev]); if (err < 0) { printk(KERN_ERR "sscape: Failed to create " - "MPU-401 device at 0x%x\n", - xport); + "MPU-401 device at 0x%lx\n", + port[dev]); goto _release_dma; } -- 1.5.2.2 ---------------------------------------------------------------------- Obdaruj swoja Walentynke ... lub siebie! Wygraj nagrody! Sprawdz >> http://link.interia.pl/f203a

2 1

[alsa-devel] [PATCH] hda: Add digital beep generator support for Realtek codecs.
by Kusanagi Kouichi 04 Feb '09

04 Feb '09

A digital beep generator can be used via input layer. Signed-off-by: Kusanagi Kouichi <slash(a)ma.neweb.ne.jp> --- sound/pci/hda/hda_beep.h | 2 +- sound/pci/hda/patch_realtek.c | 62 +++++++++++++++++++++++++++++++++++++++++ 2 files changed, 63 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index b9679f0..51bf6a5 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -39,7 +39,7 @@ struct hda_beep { int snd_hda_attach_beep_device(struct hda_codec *codec, int nid); void snd_hda_detach_beep_device(struct hda_codec *codec); #else -#define snd_hda_attach_beep_device(...) +#define snd_hda_attach_beep_device(...) 0 #define snd_hda_detach_beep_device(...) #endif #endif diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7884a4e..956b327 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -30,6 +30,7 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_beep.h" #define ALC880_FRONT_EVENT 0x01 #define ALC880_DCVOL_EVENT 0x02 @@ -3160,6 +3161,7 @@ static void alc_free(struct hda_codec *codec) alc_free_kctls(codec); kfree(spec); + snd_hda_detach_beep_device(codec); codec->spec = NULL; /* to be sure */ } @@ -4328,6 +4330,12 @@ static int patch_alc880(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC880_AUTO) setup_preset(spec, &alc880_presets[board_config]); @@ -5855,6 +5863,12 @@ static int patch_alc260(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC260_AUTO) setup_preset(spec, &alc260_presets[board_config]); @@ -7050,6 +7064,12 @@ static int patch_alc882(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC882_AUTO) setup_preset(spec, &alc882_presets[board_config]); @@ -9024,6 +9044,12 @@ static int patch_alc883(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC883_AUTO) setup_preset(spec, &alc883_presets[board_config]); @@ -10844,6 +10870,12 @@ static int patch_alc262(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC262_AUTO) setup_preset(spec, &alc262_presets[board_config]); @@ -11896,6 +11928,12 @@ static int patch_alc268(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC268_AUTO) setup_preset(spec, &alc268_presets[board_config]); @@ -12730,6 +12768,12 @@ static int patch_alc269(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC269_AUTO) setup_preset(spec, &alc269_presets[board_config]); @@ -13823,6 +13867,12 @@ static int patch_alc861(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x23); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC861_AUTO) setup_preset(spec, &alc861_presets[board_config]); @@ -14769,6 +14819,12 @@ static int patch_alc861vd(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x23); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC861VD_AUTO) setup_preset(spec, &alc861vd_presets[board_config]); @@ -16578,6 +16634,12 @@ static int patch_alc662(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC662_AUTO) setup_preset(spec, &alc662_presets[board_config]); -- 1.5.6.5

2 1

[alsa-devel] [PATCH 2.6.30] ASoC: add additional controls to the CS4270 codec driver
by Timur Tabi 04 Feb '09

04 Feb '09

Update the CS4270 codec driver to allow applications to use the mixer to control Digital Loopback, Soft Ramp, Zero Cross, Popguard, and Auto-Mute. Soft Ramp, Zero Cross, and Auto-Mute are disabled by the driver when it first initializes the hardware, but these features either don't work or interfere with normal ALSA behavior. However, they can now be re-enabled by an application if desired. Remove CONFIG_SND_SOC_CS4270_HWMUTE and always allow ASoC to control the mute bits. The driver previously and erroneously assumed that these bits control only external muting circuitry, but they also control internal muting circuitry, so they should always be used. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/codecs/Kconfig | 6 --- sound/soc/codecs/cs4270.c | 93 +++++++++++++++++++------------------------- 2 files changed, 40 insertions(+), 59 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a195303..628a591 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -67,12 +67,6 @@ config SND_SOC_AK4535 config SND_SOC_CS4270 tristate -# Cirrus Logic CS4270 Codec Hardware Mute Support -# Select if you have external muting circuitry attached to your CS4270. -config SND_SOC_CS4270_HWMUTE - bool - depends on SND_SOC_CS4270 - # Cirrus Logic CS4270 Codec VD = 3.3V Errata # Select if you are affected by the errata where the part will not function # if MCLK divide-by-1.5 is selected and VD is set to 3.3V. The driver will diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 2c79a24..cd4a9ee 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -395,17 +395,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - /* Freeze and power-down the codec */ - - ret = snd_soc_write(codec, CS4270_PWRCTL, CS4270_PWRCTL_FREEZE | - CS4270_PWRCTL_PDN_ADC | CS4270_PWRCTL_PDN_DAC | - CS4270_PWRCTL_PDN); - if (ret < 0) { - dev_err(codec->dev, "i2c write failed\n"); - return ret; - } - - /* Program the mode control register */ + /* Set the sample rate */ reg = snd_soc_read(codec, CS4270_MODE); reg &= ~(CS4270_MODE_SPEED_MASK | CS4270_MODE_DIV_MASK); @@ -417,7 +407,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, return ret; } - /* Program the format register */ + /* Set the DAI format */ reg = snd_soc_read(codec, CS4270_FORMAT); reg &= ~(CS4270_FORMAT_DAC_MASK | CS4270_FORMAT_ADC_MASK); @@ -440,42 +430,9 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, return ret; } - /* Disable auto-mute. This feature appears to be buggy, because in - some situations, auto-mute will not deactivate when it should. */ - - reg = snd_soc_read(codec, CS4270_MUTE); - reg &= ~CS4270_MUTE_AUTO; - ret = snd_soc_write(codec, CS4270_MUTE, reg); - if (ret < 0) { - dev_err(codec->dev, "i2c write failed\n"); - return ret; - } - - /* Disable automatic volume control. It's enabled by default, and - * it causes volume change commands to be delayed, sometimes until - * after playback has started. - */ - - reg = cs4270_read_reg_cache(codec, CS4270_TRANS); - reg &= ~(CS4270_TRANS_SOFT | CS4270_TRANS_ZERO); - ret = cs4270_i2c_write(codec, CS4270_TRANS, reg); - if (ret < 0) { - dev_err(codec->dev, "i2c write failed\n"); - return ret; - } - - /* Thaw and power-up the codec */ - - ret = snd_soc_write(codec, CS4270_PWRCTL, 0); - if (ret < 0) { - dev_err(codec->dev, "i2c write failed\n"); - return ret; - } - return ret; } -#ifdef CONFIG_SND_SOC_CS4270_HWMUTE /** * cs4270_mute - enable/disable the CS4270 external mute * @dai: the SOC DAI @@ -494,22 +451,23 @@ static int cs4270_mute(struct snd_soc_dai *dai, int mute) reg6 = snd_soc_read(codec, CS4270_MUTE); if (mute) - reg6 |= CS4270_MUTE_ADC_A | CS4270_MUTE_ADC_B | - CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B; + reg6 |= CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B; else - reg6 &= ~(CS4270_MUTE_ADC_A | CS4270_MUTE_ADC_B | - CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B); + reg6 &= ~(CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B); return snd_soc_write(codec, CS4270_MUTE, reg6); } -#else -#define cs4270_mute NULL -#endif /* A list of non-DAPM controls that the CS4270 supports */ static const struct snd_kcontrol_new cs4270_snd_controls[] = { SOC_DOUBLE_R("Master Playback Volume", - CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 1) + CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 1), + SOC_SINGLE("Digital Sidetone Switch", CS4270_FORMAT, 5, 1, 0), + SOC_SINGLE("Soft Ramp Switch", CS4270_TRANS, 6, 1, 0), + SOC_SINGLE("Zero Cross Switch", CS4270_TRANS, 5, 1, 0), + SOC_SINGLE("Popguard Switch", CS4270_MODE, 0, 1, 1), + SOC_SINGLE("Auto-Mute Switch", CS4270_MUTE, 5, 1, 0), + SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 0) }; /* @@ -637,6 +595,7 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, { struct snd_soc_codec *codec; struct cs4270_private *cs4270; + unsigned int reg; int ret; /* For now, we only support one cs4270 device in the system. See the @@ -702,6 +661,34 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, goto error_free_codec; } + /* Disable auto-mute. This feature appears to be buggy. In some + * situations, auto-mute will not deactivate when it should, so we want + * this feature disabled by default. An application (e.g. alsactl) can + * re-enabled it by using the controls. + */ + + reg = cs4270_read_reg_cache(codec, CS4270_MUTE); + reg &= ~CS4270_MUTE_AUTO; + ret = cs4270_i2c_write(codec, CS4270_MUTE, reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "i2c write failed\n"); + return ret; + } + + /* Disable automatic volume control. The hardware enables, and it + * causes volume change commands to be delayed, sometimes until after + * playback has started. An application (e.g. alsactl) can + * re-enabled it by using the controls. + */ + + reg = cs4270_read_reg_cache(codec, CS4270_TRANS); + reg &= ~(CS4270_TRANS_SOFT | CS4270_TRANS_ZERO); + ret = cs4270_i2c_write(codec, CS4270_TRANS, reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "i2c write failed\n"); + return ret; + } + /* Initialize the DAI. Normally, we'd prefer to have a kmalloc'd DAI * structure for each CS4270 device, but the machine driver needs to * have a pointer to the DAI structure, so for now it must be a global -- 1.5.5

2 1

[alsa-devel] [PATCH] ASoC: pxa-ssp: fix SSP port request
by Philipp Zabel 04 Feb '09

04 Feb '09

PXA2xx/3xx SSP ports start from 1, not 0. Thus, the probe function requested the wrong SSP port. Correcting this unveiled another bug where ssp_init tries to request the already-requested SSP port again. So this patch replaces the ssp_init/exit calls with their internals from mach-pxa/ssp.c, leaving out the redundant ssp_request and the unneeded IRQ request. Effectively, that leaves us with not much more than enabling/disabling the SSP clock. Signed-off-by: Philipp Zabel <philipp.zabel(a)gmail.com> --- sound/soc/pxa/pxa-ssp.c | 15 ++++++++++----- 1 files changed, 10 insertions(+), 5 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 73cb6b4..f9c6de5 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -21,6 +21,8 @@ #include <linux/clk.h> #include <linux/io.h> +#include <asm/irq.h> + #include <sound/core.h> #include <sound/pcm.h> #include <sound/initval.h> @@ -221,9 +223,9 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, int ret = 0; if (!cpu_dai->active) { - ret = ssp_init(&priv->dev, cpu_dai->id + 1, SSP_NO_IRQ); - if (ret < 0) - return ret; + priv->dev.port = cpu_dai->id + 1; + priv->dev.irq = NO_IRQ; + clk_enable(priv->dev.ssp->clk); ssp_disable(&priv->dev); } return ret; @@ -238,7 +240,7 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, if (!cpu_dai->active) { ssp_disable(&priv->dev); - ssp_exit(&priv->dev); + clk_disable(priv->dev.ssp->clk); } } @@ -751,7 +753,7 @@ static int pxa_ssp_probe(struct platform_device *pdev, if (!priv) return -ENOMEM; - priv->dev.ssp = ssp_request(dai->id, "SoC audio"); + priv->dev.ssp = ssp_request(dai->id + 1, "SoC audio"); if (priv->dev.ssp == NULL) { ret = -ENODEV; goto err_priv; -- 1.5.6.5

2 1

[alsa-devel] ASoC: UDA1380/I2S/SSP patches
by Philipp Zabel 04 Feb '09

04 Feb '09

Right now the snd-soc-magician driver in ASoC dev is severely bitrotted. I can make it compile and load again, but until now it only outputs silence (or pops and hisses as I fiddle with the uda1380 registers). For now, I'd like feedback on these patches in preparation for the magician update and switch from pxa2xx-ssp to pxa-ssp. regards Philipp

3 18

[alsa-devel] alsa license question
by Tim Blechmann 04 Feb '09

04 Feb '09

hi all, i am currently working on an alsa driver for the digigram lx6464es interface, based on the sources of the windows driver. since i would like to include some parts of the windows source code, (mainly the header files with register declarations) to the alsa driver, i have a few question: - is it ok, if these files don't match the coding style of the linux kernel? - digigram assured, that i am allowed to include source code to the linux driver. most of the files have the copyright notice: COPYRIGHT (some year) DIGIGRAM. ALL RIGHTS RESERVED. is this statement ok, or do i need to replace the `ALL RIGHTS RESERVED' with a gpl-compatible license note? - is there any repository for alsa driver documentation, provided by the manufacturer? digigram provided the windows driver for the lx6464es, mentioning that this source is `freely accessible to the GNU / ALSA community', so if there is a repository like that, the source package should probably be archived there. thanks, tim -- tim(a)klingt.org http://tim.klingt.org Avoid the world, it's just a lot of dust and drag and means nothing in the end. Jack Kerouac

3 3

[alsa-devel] Creative X-Fi USB 5.1 Driver
by Mandar Joshi 04 Feb '09

04 Feb '09

Hello, I was reading the USB Audio Specs from http://www.usb.org/developers/devclass_docs The document talks about Effect Unit Descriptors and few other descriptors.There is no mention of such descriptors in the 'lsusb -v' output for the USB X-Fi 5.1 Does this mean thet the X-Fi,Crystalizer features of the USB X-Fi are implemented purely in software? Here is a link to the lsusb output http://mndar.phpnet.us/usbxfi/files/creative.txt The document "Audio20 final.pdf" also mentions that bDeviceClass, bDeviceSubClass and bDeviceProtocol should have values 0xEF, 0x02, and 0x01 respectively but in the 'lsusb -v' output all three values are 0. Any idea why it is so? Regards Mandar Joshi

1 0

[alsa-devel] [PATCH 1/2] [EXPERIMENTAL] Add support for the latest IndigoIOx and IndigoDJx cards
by Giuliano Pochini 04 Feb '09

04 Feb '09

The following is a very beta patch and is not meant to be merged yet. It has been compile-tested only because I haven't the those cards. Hopefully I'll be able to test it myself soon. As you can see, I reused the existing IndigoIO and IndigoDJ driver because they are the same cards with a different interface. The firmware is different, though. The driver registers the card as a plain -IO or -DJ without the trailing "x" (I will fix it later if someone thinks it's important). diff -dup alsa-driver__orig/alsa-kernel/pci/echoaudio/echoaudio.h alsa-driver/alsa-kernel/pci/echoaudio/echoaudio.h --- alsa-driver__orig/alsa-kernel/pci/echoaudio/echoaudio.h 2009-02-01 15:37:54.000000000 +0100 +++ alsa-driver/alsa-kernel/pci/echoaudio/echoaudio.h 2009-02-01 21:59:38.000000000 +0100 @@ -189,6 +189,9 @@ #define INDIGO 0x0090 #define INDIGO_IO 0x00a0 #define INDIGO_DJ 0x00b0 +#define DC8 0x00c0 +#define INDIGO_IOX 0x00d0 +#define INDIGO_DJX 0x00e0 #define ECHO3G 0x0100 diff -dup alsa-driver__orig/alsa-kernel/pci/echoaudio/indigodj.c alsa-driver/alsa-kernel/pci/echoaudio/indigodj.c --- alsa-driver__orig/alsa-kernel/pci/echoaudio/indigodj.c 2009-02-01 15:37:54.000000000 +0100 +++ alsa-driver/alsa-kernel/pci/echoaudio/indigodj.c 2009-02-01 22:26:02.000000000 +0100 @@ -59,17 +59,21 @@ MODULE_FIRMWARE("ea/loader_dsp.fw"); MODULE_FIRMWARE("ea/indigo_dj_dsp.fw"); +MODULE_FIRMWARE("ea/indigo_djx_dsp.fw"); #define FW_361_LOADER 0 #define FW_INDIGO_DJ_DSP 1 +#define FW_INDIGO_DJX_DSP 2 static const struct firmware card_fw[] = { {0, "loader_dsp.fw"}, - {0, "indigo_dj_dsp.fw"} + {0, "indigo_dj_dsp.fw"}, + {0, "indigo_djx_dsp.fw"} }; static struct pci_device_id snd_echo_ids[] = { - {0x1057, 0x3410, 0xECC0, 0x00B0, 0, 0, 0}, /* Indigo DJ*/ + {0x1057, 0x3410, 0xECC0, 0x00B0, 0, 0, 0}, /* Indigo DJ */ + {0x1057, 0x3410, 0xECC0, 0x00E0, 0, 0, 0}, /* Indigo DJx */ {0,} }; diff -dup alsa-driver__orig/alsa-kernel/pci/echoaudio/indigodj_dsp.c alsa-driver/alsa-kernel/pci/echoaudio/indigodj_dsp.c --- alsa-driver__orig/alsa-kernel/pci/echoaudio/indigodj_dsp.c 2009-02-01 15:37:54.000000000 +0100 +++ alsa-driver/alsa-kernel/pci/echoaudio/indigodj_dsp.c 2009-02-01 21:53:55.000000000 +0100 @@ -37,9 +37,11 @@ static int update_vmixer_level(struct ec static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; + u16 subdev; DE_INIT(("init_hw() - Indigo DJ\n")); - if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJ)) + subdev = subdevice_id & 0xfff0; + if (snd_BUG_ON(subdev != INDIGO_DJ && subdev != INDIGO_DJX)) return -ENODEV; if ((err = init_dsp_comm_page(chip))) { @@ -50,7 +52,10 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_DJ_DSP]; + if (subdev == INDIGO_DJ) + chip->dsp_code_to_load = &card_fw[FW_INDIGO_DJ_DSP]; + else + chip->dsp_code_to_load = &card_fw[FW_INDIGO_DJX_DSP]; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff -dup alsa-driver__orig/alsa-kernel/pci/echoaudio/indigoio.c alsa-driver/alsa-kernel/pci/echoaudio/indigoio.c --- alsa-driver__orig/alsa-kernel/pci/echoaudio/indigoio.c 2009-02-01 15:37:54.000000000 +0100 +++ alsa-driver/alsa-kernel/pci/echoaudio/indigoio.c 2009-02-01 22:26:11.000000000 +0100 @@ -60,17 +60,21 @@ MODULE_FIRMWARE("ea/loader_dsp.fw"); MODULE_FIRMWARE("ea/indigo_io_dsp.fw"); +MODULE_FIRMWARE("ea/indigo_iox_dsp.fw"); #define FW_361_LOADER 0 #define FW_INDIGO_IO_DSP 1 +#define FW_INDIGO_IOX_DSP 2 static const struct firmware card_fw[] = { {0, "loader_dsp.fw"}, - {0, "indigo_io_dsp.fw"} + {0, "indigo_io_dsp.fw"}, + {0, "indigo_iox_dsp.fw"} }; static struct pci_device_id snd_echo_ids[] = { - {0x1057, 0x3410, 0xECC0, 0x00A0, 0, 0, 0}, /* Indigo IO*/ + {0x1057, 0x3410, 0xECC0, 0x00A0, 0, 0, 0}, /* Indigo IO */ + {0x1057, 0x3410, 0xECC0, 0x00D0, 0, 0, 0}, /* Indigo IOx */ {0,} }; diff -dup alsa-driver__orig/alsa-kernel/pci/echoaudio/indigoio_dsp.c alsa-driver/alsa-kernel/pci/echoaudio/indigoio_dsp.c --- alsa-driver__orig/alsa-kernel/pci/echoaudio/indigoio_dsp.c 2009-02-01 15:37:54.000000000 +0100 +++ alsa-driver/alsa-kernel/pci/echoaudio/indigoio_dsp.c 2009-02-01 21:59:05.000000000 +0100 @@ -37,9 +37,11 @@ static int update_vmixer_level(struct ec static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; + u16 subdev; DE_INIT(("init_hw() - Indigo IO\n")); - if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IO)) + subdev = subdevice_id & 0xfff0; + if (snd_BUG_ON(subdev != INDIGO_IO && subdev != INDIGO_IOX)) return -ENODEV; if ((err = init_dsp_comm_page(chip))) { @@ -50,7 +52,10 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_IO_DSP]; + if (subdev == INDIGO_IO) + chip->dsp_code_to_load = &card_fw[FW_INDIGO_IO_DSP]; + else + chip->dsp_code_to_load = &card_fw[FW_INDIGO_IOX_DSP]; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; -- Giuliano.

2 2

[alsa-devel] asoc: s3c24xx+uda1380 - some questions
by Vasily Khoruzhick 03 Feb '09

03 Feb '09

Hi, I'm trying to develop asoc driver for hp ipaq rx1950 (s3c24xx + uda1380). 1st question: where I can enable/disable uda1380 codec power? I need to control codec power for powersave reasons. I tried to add custom power function call to uda1380_set_bias_level in uda1380.c (pointer to power function is stored in codec->private_data), as it done for uda134x codec, but after enable-disable-enable cycle (for example, after resuming from suspend) driver cannot communicate with codec anymore (i2c bus is always busy), only reloading i2c driver helps. I suspect, power(0) is called in inappropriate moment, when i2c-transfer still is not finished. 2nd question: for some reason, driver does not produce any sound during first aplay invocation, only second and later aplay invocation driver produces sound. What can be the reason of this problem? Regards Vasily

3 18

[alsa-devel] [PATCH] Document tyan model for Realtek ALC262
by Tony Vroon 03 Feb '09

03 Feb '09

As just pointed out to me, the new tyan model for ALC262 was implemented but not documented. This adds the board to the list, using both its marketing name (Thunder n6650W) and its model number (S2915-E). Signed-off-by: Tony Vroon <tony(a)linx.net> diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 0f5d26b..1aede03 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -56,6 +56,7 @@ ALC262 sony-assamd Sony ASSAMD toshiba-s06 Toshiba S06 toshiba-rx1 Toshiba RX1 + tyan Tyan Thunder n6650W (S2915-E) ultra Samsung Q1 Ultra Vista model lenovo-3000 Lenovo 3000 y410 nec NEC Versa S9100

2 1

[alsa-devel] [PATCH] Add tyan model for Realtek ALC262
by Tony Vroon 03 Feb '09

03 Feb '09

The Realtek ALC262 on the Tyan Thunder n6650W (S2915-E) mainboard has a rather odd configuration template. As a result, the white AUX connector can not be used. This rewrites the default config to more accurately reflect the connector layout, colour and function. Unfortunately the black CD_IN connector, which is suspected to be widget 0x1c refuses to switch into input (0x20), instead opting to remain on 0x0. As such, no mixer controls are exposed for it. Autoswitching is implemented between the front headphone output and back line output. Signed-off-by: Tony Vroon <tony(a)linx.net> -- Takashi, this was done on top of 2.6.29-rc3. Please let me know if you'd like it rebased onto anything else instead. Unfortunately I was unable to resolve the recalcitrant widget problem, but at least the default config printed for widget 0x14 makes sense now. diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7884a4e..5759195 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -103,6 +103,7 @@ enum { ALC262_NEC, ALC262_TOSHIBA_S06, ALC262_TOSHIBA_RX1, + ALC262_TYAN, ALC262_AUTO, ALC262_MODEL_LAST /* last tag */ }; @@ -9426,6 +9427,67 @@ static struct snd_kcontrol_new alc262_benq_t31_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc262_tyan_mixer[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Aux Playback Volume", 0x0b, 0x06, HDA_INPUT), + HDA_CODEC_MUTE("Aux Playback Switch", 0x0b, 0x06, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + { } /* end */ +}; + +static struct hda_verb alc262_tyan_verbs[] = { + /* Headphone automute */ + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* P11 AUX_IN, white 4-pin connector */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, 0xe1}, + {0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, 0x93}, + {0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, 0x19}, + + {} +}; + +/* unsolicited event for HP jack sensing */ +static void alc262_tyan_automute(struct hda_codec *codec) +{ + unsigned int mute; + unsigned int present; + + snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0); + present = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0); + present = (present & 0x80000000) != 0; + if (present) { + /* mute line output on ATX panel */ + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, + HDA_AMP_MUTE, HDA_AMP_MUTE); + } else { + /* unmute line output if necessary */ + mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0); + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); + } +} + +static void alc262_tyan_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) != ALC880_HP_EVENT) + return; + alc262_tyan_automute(codec); +} + #define alc262_capture_mixer alc882_capture_mixer #define alc262_capture_alt_mixer alc882_capture_alt_mixer @@ -10534,6 +10596,7 @@ static const char *alc262_models[ALC262_MODEL_LAST] = { [ALC262_ULTRA] = "ultra", [ALC262_LENOVO_3000] = "lenovo-3000", [ALC262_NEC] = "nec", + [ALC262_TYAN] = "tyan", [ALC262_AUTO] = "auto", }; @@ -10574,6 +10637,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff7b, "Toshiba S06", ALC262_TOSHIBA_S06), SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU), + SND_PCI_QUIRK(0x10f1, 0x2915, "Tyan Thunder n6650W", ALC262_TYAN), SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA), SND_PCI_QUIRK(0x144d, 0xc039, "Samsung Q1U EL", ALC262_ULTRA), SND_PCI_QUIRK(0x144d, 0xc510, "Samsung Q45", ALC262_HIPPO), @@ -10792,6 +10856,19 @@ static struct alc_config_preset alc262_presets[] = { .unsol_event = alc262_hippo_unsol_event, .init_hook = alc262_hippo_automute, }, + [ALC262_TYAN] = { + .mixers = { alc262_tyan_mixer }, + .init_verbs = { alc262_init_verbs, alc262_tyan_verbs}, + .num_dacs = ARRAY_SIZE(alc262_dac_nids), + .dac_nids = alc262_dac_nids, + .hp_nid = 0x02, + .dig_out_nid = ALC262_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc262_modes), + .channel_mode = alc262_modes, + .input_mux = &alc262_capture_source, + .unsol_event = alc262_tyan_unsol_event, + .init_hook = alc262_tyan_automute, + }, }; static int patch_alc262(struct hda_codec *codec)

2 1

[alsa-devel] [PATCH] ASoC: email - update email addresses.
by Liam Girdwood 03 Feb '09

03 Feb '09

This just updates my email address on some drivers I'd forgotten in a previous patch. Signed-off-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- sound/soc/atmel/atmel_ssc_dai.c | 2 +- sound/soc/atmel/atmel_ssc_dai.h | 2 +- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8990.c | 3 +-- 4 files changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index c5d6790..ff0054b 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -10,7 +10,7 @@ * Based on at91-ssc.c by * Frank Mandarino <fmandarino(a)endrelia.com> * Based on pxa2xx Platform drivers by - * Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * Liam Girdwood <lrg(a)slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h index a828746..391135f 100644 --- a/sound/soc/atmel/atmel_ssc_dai.h +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -10,7 +10,7 @@ * Based on at91-ssc.c by * Frank Mandarino <fmandarino(a)endrelia.com> * Based on pxa2xx Platform drivers by - * Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * Liam Girdwood <lrg(a)slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 75d3438..d356278 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -3,7 +3,7 @@ * * Copyright (C) 2007, 2008 Wolfson Microelectronics PLC. * - * Author: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> + * Author: Liam Girdwood <lrg(a)slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 6af1d39..a2e9cc9 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -2,8 +2,7 @@ * wm8990.c -- WM8990 ALSA Soc Audio driver * * Copyright 2008 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * lg(a)opensource.wolfsonmicro.com or linux(a)wolfsonmicro.com + * Author: Liam Girdwood <lrg(a)slimlogic.co.uk> * * 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 -- 1.5.6.3

2 1

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 03 Feb '09

03 Feb '09

The following changes since commit e50a7ea0eba4f336b78b737e5ce48fba691c4919: Mark Brown (1): Merge branch 'for-2.6.29' into for-2.6.30 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Timur Tabi (2): ASoC: fix build break in CS4270 codec driver ASoC: fix message display in CS4270 codec driver sound/soc/codecs/cs4270.c | 68 ++++++++++++++++++++++++++------------------ 1 files changed, 40 insertions(+), 28 deletions(-)

2 1

[alsa-devel] Creative X-Fi USB 5.1 Driver
by Mandar Joshi 03 Feb '09

03 Feb '09

Hello, I used SniffUSB to capture the packets sent to the usb X-fi. I've uploaded it here http://mndar.phpnet.us/usbxfi/ I'm not very sure how to interpret the log. By looking at the log, can we tell if it conforms to the USB Audio spec? Regards Mandar Joshi ________________________________ Have you been here yet ? http://mndar.phpnet.us

1 0

[alsa-devel] [PATCH 2.6.30] ASoC: fix build break in CS4270 codec driver
by Timur Tabi 03 Feb '09

03 Feb '09

Fix a oversight in the CS4270 codec driver that caused a build break. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/codecs/cs4270.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index e5f5afd..7962874 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -149,7 +149,7 @@ struct cs4270_mode_ratios { u8 mclk; }; -static struct cs4270_mode_ratios[] = { +static struct cs4270_mode_ratios cs4270_mode_ratios[] = { {64, CS4270_MODE_4X, CS4270_MODE_DIV1}, #ifndef CONFIG_SND_SOC_CS4270_VD33_ERRATA {96, CS4270_MODE_4X, CS4270_MODE_DIV15}, -- 1.5.5

2 1

[alsa-devel] [2.6.29-rc3][PATCH] OMAP: ASoC: Fix spinlock misuse in omap-pcm.c
by Jarkko Nikula 03 Feb '09

03 Feb '09

From: Eero Nurkkala <ext-eero.nurkkala(a)nokia.com> omap_pcm_trigger is called also in interrupt context so CPU flags must be restored when returning. Signed-off-by: Eero Nurkkala <ext-eero.nurkkala(a)nokia.com> Acked-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- sound/soc/omap/omap-pcm.c | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index b0362df..dd3bb29 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -175,9 +175,10 @@ static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_pcm_runtime *runtime = substream->runtime; struct omap_runtime_data *prtd = runtime->private_data; + unsigned long flags; int ret = 0; - spin_lock_irq(&prtd->lock); + spin_lock_irqsave(&prtd->lock, flags); switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: @@ -195,7 +196,7 @@ static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd) default: ret = -EINVAL; } - spin_unlock_irq(&prtd->lock); + spin_unlock_irqrestore(&prtd->lock, flags); return ret; } -- 1.5.6.5

2 1

[alsa-devel] [PATCH 2.6.30] ASoC: fix message display in CS4270 codec driver
by Timur Tabi 02 Feb '09

02 Feb '09

Replace printk calls with dev_xxx calls. Set the 'dev' field of the codec and codec_dai structures so that these calls work. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/codecs/cs4270.c | 66 ++++++++++++++++++++++++++------------------ 1 files changed, 39 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 7962874..2c79a24 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -212,7 +212,7 @@ static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai, rates &= ~SNDRV_PCM_RATE_KNOT; if (!rates) { - printk(KERN_ERR "cs4270: could not find a valid sample rate\n"); + dev_err(codec->dev, "could not find a valid sample rate\n"); return -EINVAL; } @@ -253,7 +253,7 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, cs4270->mode = format & SND_SOC_DAIFMT_FORMAT_MASK; break; default: - printk(KERN_ERR "cs4270: invalid DAI format\n"); + dev_err(codec->dev, "invalid dai format\n"); ret = -EINVAL; } @@ -284,7 +284,7 @@ static int cs4270_fill_cache(struct snd_soc_codec *codec) CS4270_FIRSTREG | 0x80, CS4270_NUMREGS, cache); if (length != CS4270_NUMREGS) { - printk(KERN_ERR "cs4270: I2C read failure, addr=0x%x\n", + dev_err(codec->dev, "i2c read failure, addr=0x%x\n", i2c_client->addr); return -EIO; } @@ -340,7 +340,7 @@ static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg, if (cache[reg - CS4270_FIRSTREG] != value) { struct i2c_client *client = codec->control_data; if (i2c_smbus_write_byte_data(client, reg, value)) { - printk(KERN_ERR "cs4270: I2C write failed\n"); + dev_err(codec->dev, "i2c write failed\n"); return -EIO; } @@ -391,7 +391,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, if (i == NUM_MCLK_RATIOS) { /* We did not find a matching ratio */ - printk(KERN_ERR "cs4270: could not find matching ratio\n"); + dev_err(codec->dev, "could not find matching ratio\n"); return -EINVAL; } @@ -401,7 +401,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, CS4270_PWRCTL_PDN_ADC | CS4270_PWRCTL_PDN_DAC | CS4270_PWRCTL_PDN); if (ret < 0) { - printk(KERN_ERR "cs4270: I2C write failed\n"); + dev_err(codec->dev, "i2c write failed\n"); return ret; } @@ -413,7 +413,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, ret = snd_soc_write(codec, CS4270_MODE, reg); if (ret < 0) { - printk(KERN_ERR "cs4270: I2C write failed\n"); + dev_err(codec->dev, "i2c write failed\n"); return ret; } @@ -430,13 +430,13 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, reg |= CS4270_FORMAT_DAC_LJ | CS4270_FORMAT_ADC_LJ; break; default: - printk(KERN_ERR "cs4270: unknown format\n"); + dev_err(codec->dev, "unknown dai format\n"); return -EINVAL; } ret = snd_soc_write(codec, CS4270_FORMAT, reg); if (ret < 0) { - printk(KERN_ERR "cs4270: I2C write failed\n"); + dev_err(codec->dev, "i2c write failed\n"); return ret; } @@ -447,7 +447,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, reg &= ~CS4270_MUTE_AUTO; ret = snd_soc_write(codec, CS4270_MUTE, reg); if (ret < 0) { - printk(KERN_ERR "cs4270: I2C write failed\n"); + dev_err(codec->dev, "i2c write failed\n"); return ret; } @@ -460,7 +460,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, reg &= ~(CS4270_TRANS_SOFT | CS4270_TRANS_ZERO); ret = cs4270_i2c_write(codec, CS4270_TRANS, reg); if (ret < 0) { - printk(KERN_ERR "I2C write failed\n"); + dev_err(codec->dev, "i2c write failed\n"); return ret; } @@ -468,7 +468,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, ret = snd_soc_write(codec, CS4270_PWRCTL, 0); if (ret < 0) { - printk(KERN_ERR "cs4270: I2C write failed\n"); + dev_err(codec->dev, "i2c write failed\n"); return ret; } @@ -570,7 +570,7 @@ static int cs4270_probe(struct platform_device *pdev) /* Register PCMs */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { - printk(KERN_ERR "cs4270: failed to create PCMs\n"); + dev_err(codec->dev, "failed to create pcms\n"); return ret; } @@ -580,7 +580,7 @@ static int cs4270_probe(struct platform_device *pdev) kctrl = snd_soc_cnew(&cs4270_snd_controls[i], codec, NULL); if (!kctrl) { - printk(KERN_ERR "cs4270: error creating control '%s'\n", + dev_err(codec->dev, "error creating control '%s'\n", cs4270_snd_controls[i].name); ret = -ENOMEM; goto error_free_pcms; @@ -588,7 +588,7 @@ static int cs4270_probe(struct platform_device *pdev) ret = snd_ctl_add(codec->card, kctrl); if (ret < 0) { - printk(KERN_ERR "cs4270: error adding control '%s'\n", + dev_err(codec->dev, "error adding control '%s'\n", cs4270_snd_controls[i].name); goto error_free_pcms; } @@ -597,7 +597,7 @@ static int cs4270_probe(struct platform_device *pdev) /* And finally, register the socdev */ ret = snd_soc_init_card(socdev); if (ret < 0) { - printk(KERN_ERR "cs4270: failed to register card\n"); + dev_err(codec->dev, "failed to register card\n"); goto error_free_pcms; } @@ -643,9 +643,9 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, * comment for cs4270_codec. */ if (cs4270_codec) { - printk(KERN_ERR "cs4270: ignoring CS4270 at addr %X\n", + dev_err(&i2c_client->dev, "ignoring CS4270 at addr %X\n", i2c_client->addr); - printk(KERN_ERR "cs4270: only one CS4270 per board allowed\n"); + dev_err(&i2c_client->dev, "only one per board allowed\n"); /* Should we return something other than ENODEV here? */ return -ENODEV; } @@ -654,35 +654,35 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, ret = i2c_smbus_read_byte_data(i2c_client, CS4270_CHIPID); if (ret < 0) { - printk(KERN_ERR "cs4270: failed to read I2C at addr %X\n", + dev_err(&i2c_client->dev, "failed to read i2c at addr %X\n", i2c_client->addr); return ret; } /* The top four bits of the chip ID should be 1100. */ if ((ret & 0xF0) != 0xC0) { - printk(KERN_ERR "cs4270: device at addr %X is not a CS4270\n", + dev_err(&i2c_client->dev, "device at addr %X is not a CS4270\n", i2c_client->addr); return -ENODEV; } - printk(KERN_INFO "cs4270: found device at I2C address %X\n", + dev_info(&i2c_client->dev, "found device at i2c address %X\n", i2c_client->addr); - printk(KERN_INFO "cs4270: hardware revision %X\n", ret & 0xF); + dev_info(&i2c_client->dev, "hardware revision %X\n", ret & 0xF); /* Allocate enough space for the snd_soc_codec structure and our private data together. */ cs4270 = kzalloc(sizeof(struct cs4270_private), GFP_KERNEL); if (!cs4270) { - printk(KERN_ERR "cs4270: Could not allocate codec structure\n"); + dev_err(&i2c_client->dev, "could not allocate codec\n"); return -ENOMEM; } codec = &cs4270->codec; - cs4270_codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); + codec->dev = &i2c_client->dev; codec->name = "CS4270"; codec->owner = THIS_MODULE; codec->dai = &cs4270_dai; @@ -698,17 +698,25 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, ret = cs4270_fill_cache(codec); if (ret < 0) { - printk(KERN_ERR "cs4270: failed to fill register cache\n"); + dev_err(&i2c_client->dev, "failed to fill register cache\n"); goto error_free_codec; } + /* Initialize the DAI. Normally, we'd prefer to have a kmalloc'd DAI + * structure for each CS4270 device, but the machine driver needs to + * have a pointer to the DAI structure, so for now it must be a global + * variable. + */ + cs4270_dai.dev = &i2c_client->dev; + /* Register the DAI. If all the other ASoC driver have already * registered, then this will call our probe function, so * cs4270_codec needs to be ready. */ + cs4270_codec = codec; ret = snd_soc_register_dai(&cs4270_dai); if (ret < 0) { - printk(KERN_ERR "cs4270: failed to register DAIe\n"); + dev_err(&i2c_client->dev, "failed to register DAIe\n"); goto error_free_codec; } @@ -718,6 +726,8 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, error_free_codec: kfree(cs4270); + cs4270_codec = NULL; + cs4270_dai.dev = NULL; return ret; } @@ -733,6 +743,8 @@ static int cs4270_i2c_remove(struct i2c_client *i2c_client) struct cs4270_private *cs4270 = i2c_get_clientdata(i2c_client); kfree(cs4270); + cs4270_codec = NULL; + cs4270_dai.dev = NULL; return 0; } @@ -776,7 +788,7 @@ EXPORT_SYMBOL_GPL(soc_codec_device_cs4270); static int __init cs4270_init(void) { - printk(KERN_INFO "Cirrus Logic CS4270 ALSA SoC Codec Driver\n"); + pr_info("Cirrus Logic CS4270 ALSA SoC Codec Driver\n"); return i2c_add_driver(&cs4270_i2c_driver); } -- 1.5.5

1 0

Re: [alsa-devel] [PATCH] sscape: drop redundant fields from soundscape struct
by krzysztof.h1＠poczta.fm 02 Feb '09

02 Feb '09

Takashi Iwai napisał(a): > At Sun, 1 Feb 2009 21:12:31 +0100, > Krzysztof Helt wrote: > > > > The midi_vol is set only to 0 so it does not work > > as detection of change in midi volume. It is removed. > > It's used to cache the value. You should better to keep it, but fix > sscape_midi_put() function to remember the last > uctl->value.integer.value[0] value in this field. I know that it was supposed to cache the volume value, but is it really needed? It is not something time critical and the less code the better. > (And also better to change sscape_midi_get() to return midi_vol > value.) > Ok. Regards, Krzysztof ---------------------------------------------------------------------- Doladuj telefon! Sprawdz >> http://link.interia.pl/f2047

3 3

Re: [alsa-devel] Bug in sound-unstable-2.6 ice1724
by Vedran Miletić 02 Feb '09

02 Feb '09

On Thu, Jan 1, 2009 at 10:27 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > At Fri, 26 Dec 2008 21:34:47 +0100, > =?UTF-8?Q?Vedran_Mileti=C4=87?= wrote: >> >> On Wed, Nov 26, 2008 at 4:51 PM, Takashi Iwai <tiwai(a)suse.de> wrote: >> > Doh, I forgot to attach the patch. The below is the one. >> > >> > Please, if anyone can test it, let me know whether it works. >> > This would be nice to have in the next kernel. >> > >> > >> > thanks, >> > >> > Takashi >> > >> >> Do you still need this to be tested? I can do it. > > Yes, please. > > > Takashi > [vedran@phenomche ~]$ aplaymidi -l Port Client name Port name 14:0 Midi Through Midi Through Port-0 16:0 SB Live! 5.1 Dell OEM [SB0228] EMU10K1 MPU-401 (UART) 17:0 Emu10k1 WaveTable Emu10k1 Port 0 17:1 Emu10k1 WaveTable Emu10k1 Port 1 17:2 Emu10k1 WaveTable Emu10k1 Port 2 17:3 Emu10k1 WaveTable Emu10k1 Port 3 20:0 Terratec PHASE 22 ICE1724 MIDI [vedran@phenomche ~]$ aplaymidi -p 20:0 blah.mid [vedran@phenomche ~]$ cat /proc/asound/card1/midi0 ICE1724 MIDI Output 0 Tx bytes : 4379 Mode : native Buffer size : 4096 Avail : 4096 Input 0 Rx bytes : Works without problems. I'm using yestarday's sound-unstable git snapshot. -- Vedran Miletić

2 1

[alsa-devel] Confusion about whether snd_pcm_poll_descriptors_revents()'s revents field is a single integer or an array
by Lennart Poettering 02 Feb '09

02 Feb '09

Heya! If we look into the pcm example how snd_pcm_poll_descriptors_revents() is used then we can see that the revents parameter apparently is supposed to be a single integer. (which makes a lot of sense to me) http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html#a… However, snd_pcm_wait_nocheck() calls the same function and assumes it is a complete array! http://git.alsa-project.org/?p=alsa-lib.git;a=blob;f=src/pcm/pcm.c;h=74d1d1… So what's it now? It makes more sense to me if it would be a single fd. The problem becomnes visible if someone writes a plugin that wants wakeups from more than one fd because either snd_pcm_wait() using applications start to act weirdly because the revents are not fully initialized or snd_pcm_poll_descriptors_revents() using applications get invalid memory accesses. This has become visible in the Bluetooth plugin. I guess nobody has written a plugin that has more than one fd to sleep on yet, right? Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

3 10

[alsa-devel] Dell Latitude D830 internal speaker volume degraded on 2.6.29-rc*
by Ben Gamari 02 Feb '09

02 Feb '09

Hey all, For quite a while now, the internal speaker volume on my Dell Latitude D830 has been very soft. While usually the case can be made to resonate with all mixers at 100%, recently I have noticed a substantial decrease in output. Figuring this must be caused by a codec mis-configuration, I booted into a working kernel 2.6.27.9-134.fc10.x86_64 and collected a dump from /proc/asound/card0/codec#0. After booting back into my own vanilla kernel from master, I collected another trace. When I was looking through the diff -u of the two dumps, one particular difference seemed notable, Node 0x11 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A - Amp-Out vals: [0x7f 0x7f] - Converter: stream=5, channel=0 + Amp-Out vals: [0xff 0xff] + Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Perhaps this is significant? Regardless, I have attached both of the original dumps. Let me know if anything else is needed. Thanks, - Ben

2 3

[alsa-devel] Enumerating PCM devices
by Bankim Bhavsar 02 Feb '09

02 Feb '09

Hi Alsa developers/users, For some reason my message got truncated earlier. So here I go again. I'm using snd_device_name_hint("pcm"..) and then snd_device_name_get_hint(.."NAME") to get the NAME parameter of pcm devices. Is it true that all sound cards that support Alsa will *always* have "default" pcm device? I'm more inclined towards using "default:CARD=<vendor>" as the parameter to snd_pcm_open(). This way if the program configures hardware params for any number of channels then the appropriate PCM device will be selected on the sound card, correct? Thanks, Bankim.

2 1

[alsa-devel] snd-hda-intel option for Toshiba A200-1MB
by Vivia Nikolaidou 02 Feb '09

02 Feb '09

Dear ALSA developers, I recently installed linux on a Toshiba Satellite A200-1MB laptop. Its sound card is ALC861VD, using the hda_intel module. The problem was that the earphone jack sense wasn't working. It turns out that I had to use the Lenovo option. It would be nice if you could add this model (and probably other A200 models as well) to ALSA-Configuration.txt . Thank you, Vivia Nikolaidou

2 1

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 02 Feb '09

02 Feb '09

The following changes since commit 7393958f630ac91e591e62058f2bdb61523ec60c: Peter Ujfalusi (1): ASoC: TWL4030: Add analog loopback support are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Eero Nurkkala (1): OMAP: ASoC: Fix spinlock misuse in omap-pcm.c Grazvydas Ignotas (1): ASoC: Update OMAP3 pandora board file Mark Brown (1): Merge branch 'for-2.6.29' into for-2.6.30 Timur Tabi (2): ASoC: fix initialization order of the CS4270 codec driver ASoC: fix documentation in CS4270 codec driver sound/soc/codecs/cs4270.c | 332 ++++++++++++++++++++++++----------------- sound/soc/omap/omap-pcm.c | 5 +- sound/soc/omap/omap3pandora.c | 49 ++++--- 3 files changed, 230 insertions(+), 156 deletions(-)

2 1

[alsa-devel] ASoC update for 2.6.29
by Mark Brown 02 Feb '09

02 Feb '09

The following changes since commit ef390c0b6e3f4d2d2d43f53f4bd35e1884571a14: Misael Lopez Cruz (1): ASoC: OMAP: Initialize XCCR and RCCR registers in McBSP DAI driver are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.29 Eero Nurkkala (1): OMAP: ASoC: Fix spinlock misuse in omap-pcm.c sound/soc/omap/omap-pcm.c | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-)

2 1

[alsa-devel] [PATCH] msnd: add module description and license for the snd-msnd-lib
by Krzysztof Helt 02 Feb '09

02 Feb '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The missing module license generates warning during module loading. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/msnd/msnd.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/isa/msnd/msnd.c b/sound/isa/msnd/msnd.c index 264e082..9064544 100644 --- a/sound/isa/msnd/msnd.c +++ b/sound/isa/msnd/msnd.c @@ -700,3 +700,6 @@ int snd_msnd_pcm(struct snd_card *card, int device, } EXPORT_SYMBOL(snd_msnd_pcm); +MODULE_DESCRIPTION("Common routines for Turtle Beach Multisound drivers"); +MODULE_LICENSE("GPL"); + -- 1.5.2.2 ---------------------------------------------------------------------- Konkurs Walentynkowy! Wybierz nagrode i wygraj! Sprawdz >> http://link.interia.pl/f2039

2 1

[alsa-devel] [PATCH] sscape: drop redundant fields from soundscape struct
by Krzysztof Helt 02 Feb '09

02 Feb '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The wss_base is disguised parameter for one function. It is converted to a function parameter. The code_type is only set but never read. It is removed. The midi_vol is set only to 0 so it does not work as detection of change in midi volume. It is removed. The xport variable is alias to the port[dev]. Use the port[dev] directly to increase readability. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/sscape.c | 42 ++++++++++++++---------------------------- 1 files changed, 14 insertions(+), 28 deletions(-) diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 681e223..c6a90f0 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -135,8 +135,6 @@ enum card_type { struct soundscape { spinlock_t lock; unsigned io_base; - unsigned wss_base; - int codec_type; int ic_type; enum card_type type; struct resource *io_res; @@ -152,7 +150,6 @@ struct soundscape { spinlock_t fwlock; int hw_in_use; unsigned long midi_usage; - unsigned char midi_vol; }; #define INVALID_IRQ ((unsigned)-1) @@ -760,14 +757,9 @@ static int sscape_midi_put(struct snd_kcontrol *kctl, * and then perform another volume-related command. Perhaps the * first command is an "open" and the second command is a "close"? */ - if (s->midi_vol == ((unsigned char) uctl->value.integer. value[0] & 127)) { - change = 0; - goto __skip_change; - } change = (host_write_ctrl_unsafe(s->io_base, CMD_SET_MIDI_VOL, 100) && host_write_ctrl_unsafe(s->io_base, ((unsigned char) uctl->value.integer. value[0]) & 127, 100) && host_write_ctrl_unsafe(s->io_base, CMD_XXX_MIDI_VOL, 100)); - __skip_change: /* * Take the board out of HOST mode and back into MIDI mode ... @@ -809,12 +801,11 @@ static unsigned __devinit get_irq_config(int irq) * Perform certain arcane port-checks to see whether there * is a SoundScape board lurking behind the given ports. */ -static int __devinit detect_sscape(struct soundscape *s) +static int __devinit detect_sscape(struct soundscape *s, long wss_io) { unsigned long flags; unsigned d; int retval = 0; - int codec = s->wss_base; spin_lock_irqsave(&s->lock, flags); @@ -830,13 +821,11 @@ static int __devinit detect_sscape(struct soundscape *s) if ((d & 0x80) != 0) goto _done; - if (d == 0) { - s->codec_type = 1; + if (d == 0) s->ic_type = IC_ODIE; - } else if ((d & 0x60) != 0) { - s->codec_type = 2; + else if ((d & 0x60) != 0) s->ic_type = IC_OPUS; - } else + else goto _done; outb(0xfa, ODIE_ADDR_IO(s->io_base)); @@ -856,10 +845,10 @@ static int __devinit detect_sscape(struct soundscape *s) sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); if (s->type == SSCAPE_VIVO) - codec += 4; + wss_io += 4; /* wait for WSS codec */ for (d = 0; d < 500; d++) { - if ((inb(codec) & 0x80) == 0) + if ((inb(wss_io) & 0x80) == 0) break; spin_unlock_irqrestore(&s->lock, flags); msleep(1); @@ -1057,7 +1046,6 @@ static int __devinit create_sscape(int dev, struct snd_card *card) unsigned dma_cfg; unsigned irq_cfg; unsigned mpu_irq_cfg; - unsigned xport; struct resource *io_res; struct resource *wss_res; unsigned long flags; @@ -1077,15 +1065,15 @@ static int __devinit create_sscape(int dev, struct snd_card *card) printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); return -ENXIO; } - xport = port[dev]; /* * Grab IO ports that we will need to probe so that we * can detect and control this hardware ... */ - io_res = request_region(xport, 8, "SoundScape"); + io_res = request_region(port[dev], 8, "SoundScape"); if (!io_res) { - snd_printk(KERN_ERR "sscape: can't grab port 0x%x\n", xport); + snd_printk(KERN_ERR + "sscape: can't grab port 0x%lx\n", port[dev]); return -EBUSY; } wss_res = NULL; @@ -1112,10 +1100,9 @@ static int __devinit create_sscape(int dev, struct snd_card *card) spin_lock_init(&sscape->fwlock); sscape->io_res = io_res; sscape->wss_res = wss_res; - sscape->io_base = xport; - sscape->wss_base = wss_port[dev]; + sscape->io_base = port[dev]; - if (!detect_sscape(sscape)) { + if (!detect_sscape(sscape, wss_port[dev])) { printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", sscape->io_base); err = -ENODEV; goto _release_dma; @@ -1188,11 +1175,11 @@ static int __devinit create_sscape(int dev, struct snd_card *card) } #define MIDI_DEVNUM 0 if (sscape->type != SSCAPE_VIVO) { - err = create_mpu401(card, MIDI_DEVNUM, xport, mpu_irq[dev]); + err = create_mpu401(card, MIDI_DEVNUM, port[dev], mpu_irq[dev]); if (err < 0) { printk(KERN_ERR "sscape: Failed to create " - "MPU-401 device at 0x%x\n", - xport); + "MPU-401 device at 0x%lx\n", + port[dev]); goto _release_dma; } @@ -1204,7 +1191,6 @@ static int __devinit create_sscape(int dev, struct snd_card *card) /* * Initialize mixer */ - sscape->midi_vol = 0; host_write_ctrl_unsafe(sscape->io_base, CMD_SET_MIDI_VOL, 100); host_write_ctrl_unsafe(sscape->io_base, 0, 100); host_write_ctrl_unsafe(sscape->io_base, CMD_XXX_MIDI_VOL, 100); -- 1.5.2.2 ---------------------------------------------------------------------- Doladuj telefon! Sprawdz >> http://link.interia.pl/f2047

2 1

[alsa-devel] Recalcitrant input pin on ALC262
by Tony Vroon 02 Feb '09

02 Feb '09

Good evening, Currently I am implementing full support for the Tyan Thunder n6650W and have noticed a pin complex that advertises input, but can't seem to be set to it: Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: As you can see the pin capabilities reflect input. However, even after issuing this in the model-specific verbs: {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, the pin-ctls firmly remains 0x00. Even the python-based hda-analyzer appears unable to flip the pin-ctls over to 0x20 and reports no error messages in the GUI or in the terminal. What am I missing please? Slightly related, is it considered acceptable to modify pin defaults if they are blatantly wrong? (Reported colour, N/A vs Fixed, etc) Regards, Tony V.

2 1

[alsa-devel] [PATCH 2/2] [EXPERIMENTAL] Add support for the latest IndigoIOx and IndigoDJx cards
by Giuliano Pochini 02 Feb '09

02 Feb '09

This patch adds to alsa-firmware the firmware needed by the IndigoIOx and IndigoDJx cards. It's too long to post here inline. Please download the compressed patch at this url: http://www.webalice.it/g_pochini/alsa-firmware_indigox.patch.bz2 The patch contains two c-coded blobs and the following patch: diff -duNrp alsa-firmware__orig/echoaudio/Makefile.am alsa-firmware/echoaudio/Makefile.am --- alsa-firmware__orig/echoaudio/Makefile.am 2008-06-07 22:18:23.000000000 +0200 +++ alsa-firmware/echoaudio/Makefile.am 2009-02-01 22:15:02.000000000 +0100 @@ -26,7 +26,9 @@ firmware_files = \ 3g_asic.fw \ indigo_dsp.fw \ indigo_io_dsp.fw \ - indigo_dj_dsp.fw + indigo_iox_dsp.fw \ + indigo_dj_dsp.fw \ + indigo_djx_dsp.fw noinst_PROGRAMS = fw_writer fw_writer_SOURCES = fw_writer.c @@ -57,7 +59,9 @@ EXTRA_DIST = DSP/LoaderDSP.c \ ASIC/3G_ASIC.c \ DSP/IndigoDSP.c \ DSP/IndigoIODSP.c \ + DSP/IndigoIOxDSP.c \ DSP/IndigoDJDSP.c \ + DSP/IndigoDJxDSP.c \ fw_writer.c CLEANFILES = $(firmware_files) fw_writer @@ -97,4 +101,6 @@ $(firmware_files): fw_writer ./fw_writer ASIC/3G_ASIC.c 3g_asic.fw ./fw_writer DSP/IndigoDSP.c indigo_dsp.fw ./fw_writer DSP/IndigoIODSP.c indigo_io_dsp.fw + ./fw_writer DSP/IndigoIOxDSP.c indigo_iox_dsp.fw ./fw_writer DSP/IndigoDJDSP.c indigo_dj_dsp.fw + ./fw_writer DSP/IndigoDJxDSP.c indigo_djx_dsp.fw -- Giuliano.

1 0

[alsa-devel] work in progress: USB driver for Behringer BCD 2000
by Frank van de Pol 01 Feb '09

01 Feb '09

I'm currently working on Alsa support for the Behringer BCD 2000, a DJ control surface (http://www.behringer.com/EN/Products/BCD2000.aspx). Unfortunate this device does not act as a standard class device. Since proper technical documentation from Behringer is non-existent my work is based on analysis of the USB data stream. Current status: - driver supports midi input data for all controls/buttons that are able to send midi data (some of tje controls do not send out data but are hardwired to the analogue circuitry, eg. headphone volume etc.) still todo: - get the midi-output working to allow control of the LEDs on the control surface. - cleanup of the code (mostly remove debugging stuff) - documentation The unit has also a 4 channel 24-bit audio interface; at the moment I have little interest in trying to get this audio interface operational. If anyone else on this list has a similar device and would like to test it please let me know so I can send patches. Regards, Frank.

1 0

[alsa-devel] [PATCH] pcxhr: fix trivial typo
by Tim Blechmann 31 Jan '09

31 Jan '09

Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- sound/pci/pcxhr/pcxhr_core.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/pcxhr/pcxhr_core.h b/sound/pci/pcxhr/pcxhr_core.h index bbbd66d..be01737 100644 --- a/sound/pci/pcxhr/pcxhr_core.h +++ b/sound/pci/pcxhr/pcxhr_core.h @@ -1,7 +1,7 @@ /* * Driver for Digigram pcxhr compatible soundcards * - * low level interface with interrupt ans message handling + * low level interface with interrupt and message handling * * Copyright (c) 2004 by Digigram <alsa(a)digigram.com> * -- 1.5.6.3

2 1

[alsa-devel] [PATCH] ASoC: Update OMAP3 pandora board file
by Grazvydas Ignotas 31 Jan '09

31 Jan '09

Update pandora board file for recent TWL4030 codec changes. Also move output related snd_soc_dapm_nc_pin() calls to omap3pandora_out_init(), where they belong. Signed-off-by: Grazvydas Ignotas <notasas(a)gmail.com> --- sound/soc/omap/omap3pandora.c | 49 ++++++++++++++++++++++++---------------- 1 files changed, 29 insertions(+), 20 deletions(-) diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index fcc2f5d..fe282d4 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -143,7 +143,7 @@ static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = { }; static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = { - SND_SOC_DAPM_MIC("Mic (Internal)", NULL), + SND_SOC_DAPM_MIC("Mic (internal)", NULL), SND_SOC_DAPM_MIC("Mic (external)", NULL), SND_SOC_DAPM_LINE("Line In", NULL), }; @@ -155,16 +155,33 @@ static const struct snd_soc_dapm_route omap3pandora_out_map[] = { }; static const struct snd_soc_dapm_route omap3pandora_in_map[] = { - {"INL", NULL, "Line In"}, - {"INR", NULL, "Line In"}, - {"INL", NULL, "Mic (Internal)"}, - {"INR", NULL, "Mic (external)"}, + {"AUXL", NULL, "Line In"}, + {"AUXR", NULL, "Line In"}, + + {"MAINMIC", NULL, "Mic Bias 1"}, + {"Mic Bias 1", NULL, "Mic (internal)"}, + + {"SUBMIC", NULL, "Mic Bias 2"}, + {"Mic Bias 2", NULL, "Mic (external)"}, }; static int omap3pandora_out_init(struct snd_soc_codec *codec) { int ret; + /* All TWL4030 output pins are floating */ + snd_soc_dapm_nc_pin(codec, "OUTL"); + snd_soc_dapm_nc_pin(codec, "OUTR"); + snd_soc_dapm_nc_pin(codec, "EARPIECE"); + snd_soc_dapm_nc_pin(codec, "PREDRIVEL"); + snd_soc_dapm_nc_pin(codec, "PREDRIVER"); + snd_soc_dapm_nc_pin(codec, "HSOL"); + snd_soc_dapm_nc_pin(codec, "HSOR"); + snd_soc_dapm_nc_pin(codec, "CARKITL"); + snd_soc_dapm_nc_pin(codec, "CARKITR"); + snd_soc_dapm_nc_pin(codec, "HFL"); + snd_soc_dapm_nc_pin(codec, "HFR"); + ret = snd_soc_dapm_new_controls(codec, omap3pandora_out_dapm_widgets, ARRAY_SIZE(omap3pandora_out_dapm_widgets)); if (ret < 0) @@ -180,18 +197,11 @@ static int omap3pandora_in_init(struct snd_soc_codec *codec) { int ret; - /* All TWL4030 output pins are floating */ - snd_soc_dapm_nc_pin(codec, "OUTL"), - snd_soc_dapm_nc_pin(codec, "OUTR"), - snd_soc_dapm_nc_pin(codec, "EARPIECE"), - snd_soc_dapm_nc_pin(codec, "PREDRIVEL"), - snd_soc_dapm_nc_pin(codec, "PREDRIVER"), - snd_soc_dapm_nc_pin(codec, "HSOL"), - snd_soc_dapm_nc_pin(codec, "HSOR"), - snd_soc_dapm_nc_pin(codec, "CARKITL"), - snd_soc_dapm_nc_pin(codec, "CARKITR"), - snd_soc_dapm_nc_pin(codec, "HFL"), - snd_soc_dapm_nc_pin(codec, "HFR"), + /* Not comnnected */ + snd_soc_dapm_nc_pin(codec, "HSMIC"); + snd_soc_dapm_nc_pin(codec, "CARKITMIC"); + snd_soc_dapm_nc_pin(codec, "DIGIMIC0"); + snd_soc_dapm_nc_pin(codec, "DIGIMIC1"); ret = snd_soc_dapm_new_controls(codec, omap3pandora_in_dapm_widgets, ARRAY_SIZE(omap3pandora_in_dapm_widgets)); @@ -251,10 +261,9 @@ static int __init omap3pandora_soc_init(void) { int ret; - if (!machine_is_omap3_pandora()) { - pr_debug(PREFIX "Not OMAP3 Pandora\n"); + if (!machine_is_omap3_pandora()) return -ENODEV; - } + pr_info("OMAP3 Pandora SoC init\n"); ret = gpio_request(OMAP3_PANDORA_DAC_POWER_GPIO, "dac_power"); -- 1.5.6.3

2 1

[alsa-devel] [PATCH] ess1688: fix OPL3 port setting
by Krzysztof Helt 30 Jan '09

30 Jan '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The ess1688 driver uses the same port for PCM audio (SB compatible) and OPL3 synthesis. It is wrong so allow to choose a correct port for OPL3 synthesis. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The bug was entered as issue #282 and closed due to lack of response. However, the code is plainly wrong as MS Win95 driver reserves two separate IO areas starting from 0x220 and 0x388 for ess688 and the third 0x300 for ess1688. I have no card to test as I cut my es1688 to reuse its audio jacks few months ago. sound/isa/es1688/es1688.c | 20 +++++++++++++------- 1 files changed, 13 insertions(+), 7 deletions(-) diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index b463771..5e4d766 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -49,6 +49,7 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220,0x240,0x260 */ +static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* Usually 0x388 */ static long mpu_port[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = -1}; static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5,7,9,10 */ static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5,7,9,10 */ @@ -65,6 +66,8 @@ MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver."); module_param_array(mpu_port, long, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port # for " CRD_NAME " driver."); module_param_array(irq, int, NULL, 0444); +module_param_array(fm_port, long, NULL, 0444); +MODULE_PARM_DESC(fm_port, "FM port # for ES1688 driver."); MODULE_PARM_DESC(irq, "IRQ # for " CRD_NAME " driver."); module_param_array(mpu_irq, int, NULL, 0444); MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for " CRD_NAME " driver."); @@ -143,13 +146,16 @@ static int __devinit snd_es1688_probe(struct device *dev, unsigned int n) sprintf(card->longname, "%s at 0x%lx, irq %i, dma %i", pcm->name, chip->port, chip->irq, chip->dma8); - if (snd_opl3_create(card, chip->port, chip->port + 2, - OPL3_HW_OPL3, 0, &opl3) < 0) - dev_warn(dev, "opl3 not detected at 0x%lx\n", chip->port); - else { - error = snd_opl3_hwdep_new(opl3, 0, 1, NULL); - if (error < 0) - goto out; + if (fm_port[n] > 0 && fm_port[n] != SNDRV_AUTO_PORT) { + if (snd_opl3_create(card, fm_port[n], fm_port[n] + 2, + OPL3_HW_OPL3, 0, &opl3) < 0) + dev_warn(dev, + "opl3 not detected at 0x%lx\n", fm_port[n]); + else { + error = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (error < 0) + goto out; + } } if (mpu_irq[n] >= 0 && mpu_irq[n] != SNDRV_AUTO_IRQ && -- 1.5.2.2 ----------------------------------------------------------------------- Promocja w Speak Up. Kwartal angielskiego za darmo. 3 miesiace nauki gratis. Sprawdz teraz! >> http://link.interia.pl/f2019

3 8

[alsa-devel] [PATCH 2.6.30] ASoC: fix documentation in CS4270 codec driver
by Timur Tabi 30 Jan '09

30 Jan '09

Spruce up the documentation in the CS4270 codec. Use kerneldoc where appropriate. Fix incorrect comments. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/codecs/cs4270.c | 163 +++++++++++++++++++++++++++++---------------- 1 files changed, 105 insertions(+), 58 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index adc1150..e5f5afd 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -3,10 +3,10 @@ * * Author: Timur Tabi <timur(a)freescale.com> * - * Copyright 2007 Freescale Semiconductor, Inc. This file is licensed under - * the terms of the GNU General Public License version 2. This program - * is licensed "as is" without any warranty of any kind, whether express - * or implied. + * Copyright 2007-2009 Freescale Semiconductor, Inc. This file is licensed + * under the terms of the GNU General Public License version 2. This + * program is licensed "as is" without any warranty of any kind, whether + * express or implied. * * This is an ASoC device driver for the Cirrus Logic CS4270 codec. * @@ -111,8 +111,13 @@ struct cs4270_private { unsigned int mode; /* The mode (I2S or left-justified) */ }; -/* - * Clock Ratio Selection for Master Mode with I2C enabled +/** + * struct cs4270_mode_ratios - clock ratio tables + * @ratio: the ratio of MCLK to the sample rate + * @speed_mode: the Speed Mode bits to set in the Mode Control register for + * this ratio + * @mclk: the Ratio Select bits to set in the Mode Control register for this + * ratio * * The data for this chart is taken from Table 5 of the CS4270 reference * manual. @@ -121,31 +126,30 @@ struct cs4270_private { * It is also used by cs4270_set_dai_sysclk() to tell ALSA which sampling * rates the CS4270 currently supports. * - * Each element in this array corresponds to the ratios in mclk_ratios[]. - * These two arrays need to be in sync. - * - * 'speed_mode' is the corresponding bit pattern to be written to the + * @speed_mode is the corresponding bit pattern to be written to the * MODE bits of the Mode Control Register * - * 'mclk' is the corresponding bit pattern to be wirten to the MCLK bits of + * @mclk is the corresponding bit pattern to be wirten to the MCLK bits of * the Mode Control Register. * * In situations where a single ratio is represented by multiple speed * modes, we favor the slowest speed. E.g, for a ratio of 128, we pick * double-speed instead of quad-speed. However, the CS4270 errata states - * that Divide-By-1.5 can cause failures, so we avoid that mode where + * that divide-By-1.5 can cause failures, so we avoid that mode where * possible. * - * ERRATA: There is an errata for the CS4270 where divide-by-1.5 does not - * work if VD = 3.3V. If this effects you, select the + * Errata: There is an errata for the CS4270 where divide-by-1.5 does not + * work if Vd is 3.3V. If this effects you, select the * CONFIG_SND_SOC_CS4270_VD33_ERRATA Kconfig option, and the driver will * never select any sample rates that require divide-by-1.5. */ -static struct { +struct cs4270_mode_ratios { unsigned int ratio; u8 speed_mode; u8 mclk; -} cs4270_mode_ratios[] = { +}; + +static struct cs4270_mode_ratios[] = { {64, CS4270_MODE_4X, CS4270_MODE_DIV1}, #ifndef CONFIG_SND_SOC_CS4270_VD33_ERRATA {96, CS4270_MODE_4X, CS4270_MODE_DIV15}, @@ -162,34 +166,27 @@ static struct { /* The number of MCLK/LRCK ratios supported by the CS4270 */ #define NUM_MCLK_RATIOS ARRAY_SIZE(cs4270_mode_ratios) -/* - * Determine the CS4270 samples rates. +/** + * cs4270_set_dai_sysclk - determine the CS4270 samples rates. + * @codec_dai: the codec DAI + * @clk_id: the clock ID (ignored) + * @freq: the MCLK input frequency + * @dir: the clock direction (ignored) * - * 'freq' is the input frequency to MCLK. The other parameters are ignored. + * This function is used to tell the codec driver what the input MCLK + * frequency is. * * The value of MCLK is used to determine which sample rates are supported * by the CS4270. The ratio of MCLK / Fs must be equal to one of nine - * support values: 64, 96, 128, 192, 256, 384, 512, 768, and 1024. + * supported values - 64, 96, 128, 192, 256, 384, 512, 768, and 1024. * * This function calculates the nine ratios and determines which ones match * a standard sample rate. If there's a match, then it is added to the list - * of support sample rates. + * of supported sample rates. * * This function must be called by the machine driver's 'startup' function, * otherwise the list of supported sample rates will not be available in * time for ALSA. - * - * Note that in stand-alone mode, the sample rate is determined by input - * pins M0, M1, MDIV1, and MDIV2. Also in stand-alone mode, divide-by-3 - * is not a programmable option. However, divide-by-3 is not an available - * option in stand-alone mode. This cases two problems: a ratio of 768 is - * not available (it requires divide-by-3) and B) ratios 192 and 384 can - * only be selected with divide-by-1.5, but there is an errate that make - * this selection difficult. - * - * In addition, there is no mechanism for communicating with the machine - * driver what the input settings can be. This would need to be implemented - * for stand-alone mode to work. */ static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) @@ -230,8 +227,10 @@ static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai, return 0; } -/* - * Configure the codec for the selected audio format +/** + * cs4270_set_dai_fmt - configure the codec for the selected audio format + * @codec_dai: the codec DAI + * @format: a SND_SOC_DAIFMT_x value indicating the data format * * This function takes a bitmask of SND_SOC_DAIFMT_x bits and programs the * codec accordingly. @@ -261,8 +260,16 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, return ret; } -/* - * Pre-fill the CS4270 register cache. +/** + * cs4270_fill_cache - pre-fill the CS4270 register cache. + * @codec: the codec for this CS4270 + * + * This function fills in the CS4270 register cache by reading the register + * values from the hardware. + * + * This CS4270 registers are cached to avoid excessive I2C I/O operations. + * After the initial read to pre-fill the cache, the CS4270 never updates + * the register values, so we won't have a cache coherency problem. * * We use the auto-increment feature of the CS4270 to read all registers in * one shot. @@ -285,12 +292,17 @@ static int cs4270_fill_cache(struct snd_soc_codec *codec) return 0; } -/* - * Read from the CS4270 register cache. +/** + * cs4270_read_reg_cache - read from the CS4270 register cache. + * @codec: the codec for this CS4270 + * @reg: the register to read + * + * This function returns the value for a given register. It reads only from + * the register cache, not the hardware itself. * * This CS4270 registers are cached to avoid excessive I2C I/O operations. * After the initial read to pre-fill the cache, the CS4270 never updates - * the register values, so we won't have a cache coherncy problem. + * the register values, so we won't have a cache coherency problem. */ static unsigned int cs4270_read_reg_cache(struct snd_soc_codec *codec, unsigned int reg) @@ -303,8 +315,11 @@ static unsigned int cs4270_read_reg_cache(struct snd_soc_codec *codec, return cache[reg - CS4270_FIRSTREG]; } -/* - * Write to a CS4270 register via the I2C bus. +/** + * cs4270_i2c_write - write to a CS4270 register via the I2C bus. + * @codec: the codec for this CS4270 + * @reg: the register to write + * @value: the value to write to the register * * This function writes the given value to the given CS4270 register, and * also updates the register cache. @@ -336,11 +351,17 @@ static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg, return 0; } -/* - * Program the CS4270 with the given hardware parameters. +/** + * cs4270_hw_params - program the CS4270 with the given hardware parameters. + * @substream: the audio stream + * @params: the hardware parameters to set + * @dai: the SOC DAI (ignored) * - * The .ops functions are used to provide board-specific data, like - * input frequencies, to this driver. This function takes that information, + * This function programs the hardware with the values provided. + * Specifically, the sample rate and the data format. + * + * The .ops functions are used to provide board-specific data, like input + * frequencies, to this driver. This function takes that information, * combines it with the hardware parameters provided, and programs the * hardware accordingly. */ @@ -455,8 +476,10 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, } #ifdef CONFIG_SND_SOC_CS4270_HWMUTE -/* - * Set the CS4270 external mute +/** + * cs4270_mute - enable/disable the CS4270 external mute + * @dai: the SOC DAI + * @mute: 0 = disable mute, 1 = enable mute * * This function toggles the mute bits in the MUTE register. The CS4270's * mute capability is intended for external muting circuitry, so if the @@ -490,7 +513,7 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = { }; /* - * Global variable to store codec for the ASoC probe function. + * cs4270_codec - global variable to store codec for the ASoC probe function * * If struct i2c_driver had a private_data field, we wouldn't need to use * cs4270_codec. This is the only way to pass the codec structure from @@ -527,8 +550,12 @@ struct snd_soc_dai cs4270_dai = { }; EXPORT_SYMBOL_GPL(cs4270_dai); -/* - * ASoC probe function +/** + * cs4270_probe - ASoC probe function + * @pdev: platform device + * + * This function is called when ASoC has all the pieces it needs to + * instantiate a sound driver. */ static int cs4270_probe(struct platform_device *pdev) { @@ -582,6 +609,12 @@ error_free_pcms: return ret; } +/** + * cs4270_remove - ASoC remove function + * @pdev: platform device + * + * This function is the counterpart to cs4270_probe(). + */ static int cs4270_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); @@ -591,14 +624,13 @@ static int cs4270_remove(struct platform_device *pdev) return 0; }; -/* - * Initialize the I2C interface of the CS4270 - * - * This function is called for whenever the I2C subsystem finds a device - * at a particular address. +/** + * cs4270_i2c_probe - initialize the I2C interface of the CS4270 + * @i2c_client: the I2C client object + * @id: the I2C device ID (ignored) * - * Note: snd_soc_new_pcms() must be called before this function can be called, - * because of snd_ctl_add(). + * This function is called whenever the I2C subsystem finds a device that + * matches the device ID given via a prior call to i2c_add_driver(). */ static int cs4270_i2c_probe(struct i2c_client *i2c_client, const struct i2c_device_id *id) @@ -690,6 +722,12 @@ error_free_codec: return ret; } +/** + * cs4270_i2c_remove - remove an I2C device + * @i2c_client: the I2C client object + * + * This function is the counterpart to cs4270_i2c_probe(). + */ static int cs4270_i2c_remove(struct i2c_client *i2c_client) { struct cs4270_private *cs4270 = i2c_get_clientdata(i2c_client); @@ -699,12 +737,21 @@ static int cs4270_i2c_remove(struct i2c_client *i2c_client) return 0; } +/* + * cs4270_id - I2C device IDs supported by this driver + */ static struct i2c_device_id cs4270_id[] = { {"cs4270", 0}, {} }; MODULE_DEVICE_TABLE(i2c, cs4270_id); +/* + * cs4270_i2c_driver - I2C device identification + * + * This structure tells the I2C subsystem how to identify and support a + * given I2C device type. + */ static struct i2c_driver cs4270_i2c_driver = { .driver = { .name = "cs4270", -- 1.5.5

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[alsa-devel] HP DV4 1155 SE - No Sound through speakers...
by Tim Barnette 30 Jan '09

30 Jan '09

Hi all, I have an HP DV4 1155-SE, I've been unable to get sound working through the speakers, although it does work through the headhpones. I ran alsa-.upgrade which downloaded compiled, and installed 1.0.19 - still no luck. I ran alsa-info, output here: http://www.alsa-project.org/db/?f=844b7ee6468067460e17b7fc694e49052072a26f I filed a bug, here: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4374 Any ideas?

2 1

[alsa-devel] Firmware
by soundprizm＠gmail.com 30 Jan '09

30 Jan '09

Is ALSA Firmware written by alsa developers or obtained from the hardware's manufacturer? If it is obtained from the manufacturer, what is exactly required to have the .ihx files created, or is that the original firmware file format?

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[alsa-devel] Edirol UA-20 in advanced mode
by nick83ola 30 Jan '09

30 Jan '09

Hello to all I have an old edirol UA-20 that not work with advanced driver, or perhaps I see it, i can send midi in and out from it but audio don't work. I have ubuntu intrepid and Jaunty (all the same problem!!!) The sound card works perfectly with advanced driver off, but in this mode is useless (no midi/no low latency audio/no 24 bit) here's the output of some command damnick@damnick-nb:~$ lsusb Bus 005 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 004 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub Bus 003 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub Bus 002 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub Bus 001 Device 003: ID 0582:0025 Roland Corp. EDIROL UA-20 Bus 001 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub damnick@damnick-nb:~$ sudo lsusb -v -d 0582:0025 [sudo] password for damnick: Bus 001 Device 003: ID 0582:0025 Roland Corp. EDIROL UA-20 Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 255 bMaxPacketSize0 8 idVendor 0x0582 Roland Corp. idProduct 0x0025 EDIROL UA-20 bcdDevice 1.01 iManufacturer 1 Roland iProduct 2 EDIROL UA-20 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 322 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 170mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 0a 24 01 00 01 34 00 02 01 02 ** UNRECOGNIZED: 0c 24 02 01 01 01 00 02 03 00 00 00 ** UNRECOGNIZED: 09 24 03 03 04 03 00 01 00 ** UNRECOGNIZED: 0c 24 02 04 01 06 00 02 03 00 00 00 ** UNRECOGNIZED: 09 24 03 07 01 01 00 04 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 07 24 01 01 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 03 18 01 44 ac 00 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0120 1x 288 bytes bInterval 1 bRefresh 0 bSynchAddress 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 07 24 01 07 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 03 18 01 44 ac 00 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0120 1x 288 bytes bInterval 1 bRefresh 0 bSynchAddress 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 07 24 01 00 01 25 00 ** UNRECOGNIZED: 06 24 02 01 10 00 ** UNRECOGNIZED: 06 24 02 02 20 00 ** UNRECOGNIZED: 09 24 03 01 30 01 20 01 00 ** UNRECOGNIZED: 09 24 03 02 40 01 10 01 00 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 0 bRefresh 0 bSynchAddress 0 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 0 bRefresh 0 bSynchAddress 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 07 24 01 00 01 25 00 ** UNRECOGNIZED: 06 24 02 01 10 00 ** UNRECOGNIZED: 06 24 02 02 20 00 ** UNRECOGNIZED: 09 24 03 01 30 01 20 01 00 ** UNRECOGNIZED: 09 24 03 02 40 01 10 01 00 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 0 bRefresh 0 bSynchAddress 0 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 1 bRefresh 0 bSynchAddress 0 Device Status: 0x0001 Self Powered damnick@damnick-nb:~$ aplay -l **** Lista di PLAYBACK dispositivi hardware **** scheda 0: ICH6 [Intel ICH6], dispositivo 0: Intel ICH [Intel ICH6] Sottoperiferiche: 1/1 Sottoperiferica #0: subdevice #0 scheda 0: ICH6 [Intel ICH6], dispositivo 4: Intel ICH - IEC958 [Intel ICH6 - IEC958] Sottoperiferiche: 1/1 Sottoperiferica #0: subdevice #0 scheda 1: UA20 [UA-20], dispositivo 0: USB Audio [USB Audio] Sottoperiferiche: 1/1 Sottoperiferica #0: subdevice #0 damnick@damnick-nb:~$ dmesg|tail [ 123.661964] [drm] Num pipes: 1 [ 151.700439] ieee80211_crypt: registered algorithm 'TKIP' [ 155.428103] CPU0 attaching NULL sched-domain. [ 155.428227] CPU0 attaching NULL sched-domain. [ 194.768219] ACPI: EC: non-query interrupt received, switching to interrupt mode [ 194.771985] ACPI: EC: GPE storm detected, transactions will use polling mode [ 1247.360077] usb 1-1: USB disconnect, address 2 [ 1257.952020] usb 1-1: new full speed USB device using uhci_hcd and address 3 [ 1258.228810] usb 1-1: configuration #1 chosen from 1 choice [ 1287.914930] cannot submit datapipe for urb 0, error -90: internal error damnick@damnick-nb:~$ aplay -D plughw:UA20 /usr/share/sounds/question.wav Riproduzione in corso WAVE '/usr/share/sounds/question.wav' : Signed 16 bit Little Endian, Frequenza 44100 Hz, Mono ^C Interrotto dal segnale Interrupt... and no sound.... jack hangs when I try to open the device.... I have seen an open discussion here http://lists.linuxaudio.org/pipermail/linux-audio-user/2005-November/028696… I have the same output with lsusb... but the thread ends here and no fixes!!! Anyone can help me???? Please??? Nicola Lunghi

2 1

[alsa-devel] [PATCH 2.6.30] ASoC: fix initialization order of the CS4270 codec driver
by Timur Tabi 30 Jan '09

30 Jan '09

ASoC codec drivers typically serve two masters: the I2C bus and ASoC itself. When a codec driver registers with ASoC, a probe function is called. Most codec drivers call ASoC first, and then register with the I2C bus in the ASoC probe function. However, in order to support multiple codecs on one board, it's easier if the codec driver is probed via the I2C bus first. This is because the call to i2c_add_driver() can result in the I2C probe function being called multiple times - once for each codec. In the current design, the driver registers once with ASoC, and in the ASoC probe function, it calls i2c_add_driver(). The results in the I2C probe function being called multiple times before the driver can register with ASoC again. The new design has the driver call i2c_add_driver() first. In the I2C probe function, the driver registers with ASoC. This allows the ASoC probe function to be called once per I2C device. Also add code to check if the I2C probe function is called more than once, since that is not supported with the current ASoC design. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/codecs/cs4270.c | 177 +++++++++++++++++++++++++-------------------- 1 files changed, 97 insertions(+), 80 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 21253b4..adc1150 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -490,21 +490,17 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = { }; /* - * Global variable to store socdev for i2c probe function. + * Global variable to store codec for the ASoC probe function. * * If struct i2c_driver had a private_data field, we wouldn't need to use - * cs4270_socdec. This is the only way to pass the socdev structure to - * cs4270_i2c_probe(). - * - * The real solution to cs4270_socdev is to create a mechanism - * that maps I2C addresses to snd_soc_device structures. Perhaps the - * creation of the snd_soc_device object should be moved out of - * cs4270_probe() and into cs4270_i2c_probe(), but that would make this - * driver dependent on I2C. The CS4270 supports "stand-alone" mode, whereby - * the chip is *not* connected to the I2C bus, but is instead configured via - * input pins. + * cs4270_codec. This is the only way to pass the codec structure from + * cs4270_i2c_probe() to cs4270_probe(). Unfortunately, there is no good + * way to synchronize these two functions. cs4270_i2c_probe() can be called + * multiple times before cs4270_probe() is called even once. So for now, we + * also only allow cs4270_i2c_probe() to be run once. That means that we do + * not support more than one cs4270 device in the system, at least for now. */ -static struct snd_soc_device *cs4270_socdev; +static struct snd_soc_codec *cs4270_codec; struct snd_soc_dai cs4270_dai = { .name = "cs4270", @@ -532,6 +528,70 @@ struct snd_soc_dai cs4270_dai = { EXPORT_SYMBOL_GPL(cs4270_dai); /* + * ASoC probe function + */ +static int cs4270_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = cs4270_codec; + unsigned int i; + int ret; + + /* Connect the codec to the socdev. snd_soc_new_pcms() needs this. */ + socdev->card->codec = codec; + + /* Register PCMs */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "cs4270: failed to create PCMs\n"); + return ret; + } + + /* Add the non-DAPM controls */ + for (i = 0; i < ARRAY_SIZE(cs4270_snd_controls); i++) { + struct snd_kcontrol *kctrl; + + kctrl = snd_soc_cnew(&cs4270_snd_controls[i], codec, NULL); + if (!kctrl) { + printk(KERN_ERR "cs4270: error creating control '%s'\n", + cs4270_snd_controls[i].name); + ret = -ENOMEM; + goto error_free_pcms; + } + + ret = snd_ctl_add(codec->card, kctrl); + if (ret < 0) { + printk(KERN_ERR "cs4270: error adding control '%s'\n", + cs4270_snd_controls[i].name); + goto error_free_pcms; + } + } + + /* And finally, register the socdev */ + ret = snd_soc_init_card(socdev); + if (ret < 0) { + printk(KERN_ERR "cs4270: failed to register card\n"); + goto error_free_pcms; + } + + return 0; + +error_free_pcms: + snd_soc_free_pcms(socdev); + + return ret; +} + +static int cs4270_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + + return 0; +}; + +/* * Initialize the I2C interface of the CS4270 * * This function is called for whenever the I2C subsystem finds a device @@ -543,17 +603,27 @@ EXPORT_SYMBOL_GPL(cs4270_dai); static int cs4270_i2c_probe(struct i2c_client *i2c_client, const struct i2c_device_id *id) { - struct snd_soc_device *socdev = cs4270_socdev; struct snd_soc_codec *codec; struct cs4270_private *cs4270; - int i; - int ret = 0; + int ret; + + /* For now, we only support one cs4270 device in the system. See the + * comment for cs4270_codec. + */ + if (cs4270_codec) { + printk(KERN_ERR "cs4270: ignoring CS4270 at addr %X\n", + i2c_client->addr); + printk(KERN_ERR "cs4270: only one CS4270 per board allowed\n"); + /* Should we return something other than ENODEV here? */ + return -ENODEV; + } /* Verify that we have a CS4270 */ ret = i2c_smbus_read_byte_data(i2c_client, CS4270_CHIPID); if (ret < 0) { - printk(KERN_ERR "cs4270: failed to read I2C\n"); + printk(KERN_ERR "cs4270: failed to read I2C at addr %X\n", + i2c_client->addr); return ret; } /* The top four bits of the chip ID should be 1100. */ @@ -575,7 +645,7 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, return -ENOMEM; } codec = &cs4270->codec; - socdev->card->codec = codec; + cs4270_codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); @@ -600,50 +670,20 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, goto error_free_codec; } - /* Register PCMs */ - - ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + /* Register the DAI. If all the other ASoC driver have already + * registered, then this will call our probe function, so + * cs4270_codec needs to be ready. + */ + ret = snd_soc_register_dai(&cs4270_dai); if (ret < 0) { - printk(KERN_ERR "cs4270: failed to create PCMs\n"); + printk(KERN_ERR "cs4270: failed to register DAIe\n"); goto error_free_codec; } - /* Add the non-DAPM controls */ - - for (i = 0; i < ARRAY_SIZE(cs4270_snd_controls); i++) { - struct snd_kcontrol *kctrl; - - kctrl = snd_soc_cnew(&cs4270_snd_controls[i], codec, NULL); - if (!kctrl) { - printk(KERN_ERR "cs4270: error creating control '%s'\n", - cs4270_snd_controls[i].name); - ret = -ENOMEM; - goto error_free_pcms; - } - - ret = snd_ctl_add(codec->card, kctrl); - if (ret < 0) { - printk(KERN_ERR "cs4270: error adding control '%s'\n", - cs4270_snd_controls[i].name); - goto error_free_pcms; - } - } - - /* Initialize the SOC device */ - - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "cs4270: failed to register card\n"); - goto error_free_pcms;; - } - - i2c_set_clientdata(i2c_client, socdev); + i2c_set_clientdata(i2c_client, cs4270); return 0; -error_free_pcms: - snd_soc_free_pcms(socdev); - error_free_codec: kfree(cs4270); @@ -652,11 +692,8 @@ error_free_codec: static int cs4270_i2c_remove(struct i2c_client *i2c_client) { - struct snd_soc_device *socdev = i2c_get_clientdata(i2c_client); - struct snd_soc_codec *codec = socdev->card->codec; - struct cs4270_private *cs4270 = codec->private_data; + struct cs4270_private *cs4270 = i2c_get_clientdata(i2c_client); - snd_soc_free_pcms(socdev); kfree(cs4270); return 0; @@ -679,26 +716,6 @@ static struct i2c_driver cs4270_i2c_driver = { }; /* - * ASoC probe function - * - * This function is called when the machine driver calls - * platform_device_add(). - */ -static int cs4270_probe(struct platform_device *pdev) -{ - cs4270_socdev = platform_get_drvdata(pdev);; - - return i2c_add_driver(&cs4270_i2c_driver); -} - -static int cs4270_remove(struct platform_device *pdev) -{ - i2c_del_driver(&cs4270_i2c_driver); - - return 0; -} - -/* * ASoC codec device structure * * Assign this variable to the codec_dev field of the machine driver's @@ -714,13 +731,13 @@ static int __init cs4270_init(void) { printk(KERN_INFO "Cirrus Logic CS4270 ALSA SoC Codec Driver\n"); - return snd_soc_register_dai(&cs4270_dai); + return i2c_add_driver(&cs4270_i2c_driver); } module_init(cs4270_init); static void __exit cs4270_exit(void) { - snd_soc_unregister_dai(&cs4270_dai); + i2c_del_driver(&cs4270_i2c_driver); } module_exit(cs4270_exit); -- 1.5.5

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[alsa-devel] Very faint sound on Dell Latitude E6500
by Laurent Pinchart 30 Jan '09

30 Jan '09

Hi everybody, I've received a brand new Dell Latitude E6500 and found out that the sound card wasn't exactly well supported by ALSA. Sound is barely audible (so faint that I'm not even sure if the proper sound is output or if I just get garbage), with some loud noise here and there that seem to correspond to the saturated audio data parts. The sound card is an HDA-Intel with an IDT 92HD71B7X codec. Hardware information can be found at http://www.alsa-project.org/db/?f=30cac30e5b1ddd9c842481a830c250cc3e593b0d. A bug report including the Windows driver .ini file is available at https://bugtrack.alsa-project.org/alsa-bug/bug_view_advanced_page.php?bug_i…. When loading the snd_hda_intel driver compiled with verbose debug mode, the following messages are printed to the kernel log. HDA Intel 0000:00:1b.0: PCI INT A -> GSI 21 (level, low) -> IRQ 21 HDA Intel 0000:00:1b.0: setting latency timer to 64 ALSA sound/pci/hda/hda_intel.c:2209: chipset global capabilities = 0x4401 ALSA sound/pci/hda/hda_intel.c:775: codec_mask = 0x5 ALSA sound/pci/hda/hda_codec.c:2367: hda_codec: model 'dell-m4-1' is selected for config 1028:24f (unknown Dell) ALSA sound/pci/hda/patch_sigmatel.c:2247: hda_codec: pin nid 0a pin config 0421101f ALSA sound/pci/hda/patch_sigmatel.c:2247: hda_codec: pin nid 0b pin config 04a11221 ALSA sound/pci/hda/patch_sigmatel.c:2247: hda_codec: pin nid 0c pin config 40f000f0 ALSA sound/pci/hda/patch_sigmatel.c:2247: hda_codec: pin nid 0d pin config 90170110 ALSA sound/pci/hda/patch_sigmatel.c:2247: hda_codec: pin nid 0e pin config 23a1902e ALSA sound/pci/hda/patch_sigmatel.c:2247: hda_codec: pin nid 0f pin config 23014250 ALSA sound/pci/hda/patch_sigmatel.c:2247: hda_codec: pin nid 14 pin config 40f000f0 ALSA sound/pci/hda/patch_sigmatel.c:2247: hda_codec: pin nid 18 pin config 90a000f0 ALSA sound/pci/hda/patch_sigmatel.c:2247: hda_codec: pin nid 19 pin config 40f000f0 ALSA sound/pci/hda/patch_sigmatel.c:2247: hda_codec: pin nid 1e pin config 4f0000f0 ALSA sound/pci/hda/patch_sigmatel.c:2247: hda_codec: pin nid 1f pin config 4f0000f0 ALSA sound/pci/hda/hda_codec.c:3055: autoconfig: line_outs=1 (0xf/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:3059: speaker_outs=1 (0xd/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:3063: hp_outs=1 (0xa/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:3064: mono: mono_out=0x0 ALSA sound/pci/hda/hda_codec.c:3072: inputs: mic=0xb, fmic=0xe, line=0x0, fline=0x0, cd=0x0, aux=0x0 ALSA sound/pci/hda/patch_sigmatel.c:2693: stac92xx_add_dyn_out_pins: total dac count=2 ALSA sound/pci/hda/hda_generic.c:679: hda_generic: no proper input path found ALSA sound/pci/hda/hda_generic.c:284: Skip Digital OUT node 2 ALSA sound/pci/hda/hda_generic.c:427: hda_generic: no proper output path found ALSA sound/pci/hda/hda_generic.c:1022: hda_generic: no PCM found ALSA sound/pci/hda/hda_codec.c:1127: Cannot find slave Surround Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:1127: Cannot find slave Center Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:1127: Cannot find slave LFE Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:1127: Cannot find slave Side Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:1127: Cannot find slave Speaker Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:1127: Cannot find slave External Speaker Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:1127: Cannot find slave Speaker2 Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:1127: Cannot find slave Surround Playback Switch, skipped ALSA sound/pci/hda/hda_codec.c:1127: Cannot find slave Center Playback Switch, skipped ALSA sound/pci/hda/hda_codec.c:1127: Cannot find slave LFE Playback Switch, skipped ALSA sound/pci/hda/hda_codec.c:1127: Cannot find slave Side Playback Switch, skipped ALSA sound/pci/hda/hda_codec.c:1127: Cannot find slave Speaker Playback Switch, skipped ALSA sound/pci/hda/hda_codec.c:1127: Cannot find slave External Speaker Playback Switch, skipped ALSA sound/pci/hda/hda_codec.c:1127: Cannot find slave Speaker2 Playback Switch, skipped ALSA sound/pci/hda/hda_codec.c:1127: Cannot find slave IEC958 Playback Switch, skipped I tried forcing the model to dell-m4-1, dell-m4-2, dell-m4-3 and ref without any luck. I also tried upgrading to alsa-driver 1.0.19 and results were identical. Could anyone help me ? I can patch and recompile the driver is required. Best regards, Laurent Pinchart

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[alsa-devel] kerneldoc for sound drivers?
by Timur Tabi 30 Jan '09

30 Jan '09

I'm fixing the comments in my driver (sound/soc/codecs/cs4270.c) to conform to kerneldoc standards. However, when I build "make htmldocs", my comments aren't processed. I presume the make rules don't know to look into the sound/soc/codecs directory. How do I fix this? -- Timur Tabi Linux kernel developer at Freescale

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[alsa-devel] Usb driver debug
by nick83ola 30 Jan '09

30 Jan '09

anyone can give me information about how i can debug the driver for my usb edirol ua-20 ?(that not works in advanced mode...sigh sob) i have downloaded the kernel source and locked in sound/usb any examples about how to debug/develop usb linux drivers??? Please I need it!!!!! Nicola Lunghi -- www.nicolalunghi.it http://www.cerchiopisa3.netsons.org P.S. Le informazioni trasmesse attraverso la presente comunicazione sono di sclusiva spettanza dell'effettivo destinatario. Nel caso in cui le stesse raggiungessero, per qualunque motivo, soggetti non interessati, questi ultimi vorranno darne immediata notizia al mittente. In ogni caso, eventuali soggetti diversi dai legittimi destinatari della presente comunicazione e dei dati contenuti negli allegati, possono essere sanzionati ai sensi del T.U. sul trattamento dei dati personali d.lgs. 196/2003, sia ai sensi dell'art. 616 del Codice Penale che disciplina la violazione del segreto sulla corrispondenza.

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[alsa-devel] Creative X-Fi USB 5.1 Driver
by Mandar Joshi 30 Jan '09

30 Jan '09

Hello, I walked through the code for snd-usb-audio and have a fair idea about the structure of the driver. I also went through the "writing-an-alsa-driver" article on http://www.alsa-project.org Here is the stuff I've done so far http://mndar.phpnet.us/usbxfi/ .The link contains the "lsusb -v" output for the Creative USB X-Fi 5.1 . The xfi-0.0.0.0.tar.bz2 has the basic structure of an alsa driver (copied from the caiaq usb driver). It presently just loads the module when the card is connected. Output in /var/log/messages ------------------------------------------- Jan 30 10:46:29 mandar Found Creative USB X-Fi 5.1!! Jan 30 10:46:29 mandar 0 041e:3042 Jan 30 10:46:29 mandar Found Creative USB X-Fi 5.1!! Jan 30 10:46:29 mandar 0 041e:3042 Jan 30 10:46:29 mandar Found Creative USB X-Fi 5.1!! Jan 30 10:46:29 mandar 0 041e:3042 Jan 30 10:46:29 mandar usbcore: registered new interface driver snd-usb-xfi -------------------------------------------------------------------------------- As you can see, the probe is being called 3 times. Can someone please go through the source and let me know if I'm headed in the right direction? I'm a newbie to alsa driver development and would really appreciate any input from you guys. Also, please have a look the creative.txt file in the link. Its the 'lsusb -v' output for the card. How exactly do I interpret and use this information? Regards Mandar Joshi

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[alsa-devel] No digital output on HP Pavilion dv6936us (intel_hda ALC268), also volume low--help me hack?
by Jens Jorgensen 30 Jan '09

30 Jan '09

Hello, I've got sound working on my laptop but there are two problems I'd like to solve. One is the volume is kind of low. Not so low I'm positive something is wrong, but it seems like it isn't right. Second, and more interesting, is this laptop seems to digital audio output that goes over the HDMI connector. Or, I should say, in Windows XP (installing that was another adventure I would have preferred not to have done) The mixer has a control to enable S/PDIF output. I'm a (non-kernel) c++ developer by trade and I've got a local kernel tree from git and should be able to manage pulling the alsa git as well, so if some alsa person could guide me along in hacking the driver to help me try to get this working that would be great. I ran alsa-info.sh on my box, here's the URL: http://www.alsa-project.org/db/?f=1b5a45d1336cee176dbd99e725a56f7ad4e20b42 -- Jens B. Jorgensen jbj1(a)ultraemail.net

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[alsa-devel] Can DMA address, periods, buffer size, etc change after hw_params is called?
by Hans-Christian Egtvedt 30 Jan '09

30 Jan '09

Hi, I wonder if I must take care handling changes in DMA address, number of periods and period size in the prepare callback? Or are these locked after the first time prepare is called? The reason I ask is because I'm developing a new ALSA driver and I prepare a cyclic DMA transfer in the first prepare callback. Additional calls to prepare will not do anything DMA stuff. The cyclic DMA transfer is then released in the close callback. Should I after the open and hw_params calls be ready to handle any change at all regarding DMA buffer address, periods and/or number of periods? I hope I made sense in this email (-: -- Best regards, Hans-Christian Egtvedt

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[alsa-devel] [PATCH 0/1] ASoC: TWL4030: Analog bypass implementation - try2
by Peter Ujfalusi 29 Jan '09

29 Jan '09

Hello, Second try for the analog bypass implementation. Since the first try most of the things has been turned around - I tried to follow Mark's advice, I hope I managed well enough: [1] New power, codec state handling The codec will be powered at all time, except if the codec's bias level has been set to SND_SOC_BIAS_OFF via the twl4030_set_bias_level call. [2] Keeping the power consumption on the lowest level When the codec is in SND_SOC_BIAS_ON state (playback/recording), there is nothing can be done, the amplifiers are powered, PLL is enabled. When the codec enters to SND_SOC_BIAS_STANDBY state, two thing can happen: - If none of the bypass switches are enabled, than the gain controls for the outputs are forced to power-down mode and the PLL is disabled - If one of the bypass switches are enabled, than the gain controls will be kept in their user configured state and the PLL stays enabled. When the codec is in SND_SOC_BIAS_STANDBY (gains and PLL are disabled), change in the bypass config will trigger an event and if it is needed the gains and PLL will be enabled. Switching all bypass off will switch the gains and PLL off. With the this implementation, there is no visible difference in power consumption compared to the previous, codec off way. So when the codec is in SND_SOC_BIAS_STANDBY mode and all of the bypasses are disabled, the additional power consumption is minimal, if any. --- Peter Ujfalusi (1): ASoC: TWL4030: Add analog loopback support sound/soc/codecs/twl4030.c | 212 +++++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/twl4030.h | 15 +++ 2 files changed, 214 insertions(+), 13 deletions(-)

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[alsa-devel] [PATCH] ALSA: hda - make alc882_auto_init_input_src aware of selectors
by Herton Ronaldo Krzesinski 29 Jan '09

29 Jan '09

In the case of having a selector instead of mixer while initing input sources, the case that happens with ALC889, we must select instead of muting input. Problem was found while testing with hda-emu. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> --- sound/pci/hda/patch_realtek.c | 21 ++++++++++++++++++--- 1 files changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 56adbb8..31a6d40 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6924,18 +6924,21 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec) static void alc882_auto_init_input_src(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - const struct hda_input_mux *imux = spec->input_mux; int c; for (c = 0; c < spec->num_adc_nids; c++) { hda_nid_t conn_list[HDA_MAX_NUM_INPUTS]; hda_nid_t nid = spec->capsrc_nids[c]; + unsigned int mux_idx; + const struct hda_input_mux *imux; int conns, mute, idx, item; conns = snd_hda_get_connections(codec, nid, conn_list, ARRAY_SIZE(conn_list)); if (conns < 0) continue; + mux_idx = c >= spec->num_mux_defs ? 0 : c; + imux = &spec->input_mux[mux_idx]; for (idx = 0; idx < conns; idx++) { /* if the current connection is the selected one, * unmute it as default - otherwise mute it @@ -6948,8 +6951,20 @@ static void alc882_auto_init_input_src(struct hda_codec *codec) break; } } - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, mute); + /* check if we have a selector or mixer + * we could check for the widget type instead, but + * just check for Amp-In presence (in case of mixer + * without amp-in there is something wrong, this + * function shouldn't be used or capsrc nid is wrong) + */ + if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + mute); + else if (mute != AMP_IN_MUTE(idx)) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, + idx); } } } -- 1.6.1

2 1

[alsa-devel] CS4232 Playback bug on Tecra 500
by Stanley Pinchak 29 Jan '09

29 Jan '09

I have been running into the problem briefly mentioned at the wiki entry http://alsa.opensrc.org/index.php/Toshiba_Tecra_500CDT. Playback using the CS4232 driver in native or oss emulation mode fails randomly. Once the failure occurs, subsequent attempts to playback stop shortly after opening the device and tickling the speakers (perhaps the buffer gets filled once). Under heavy network load, the initial failure will occur rather soon, within 15 minutes or so of playback. When I am playing songs from the hard drive, the failure takes a very long time, and when I use oss emulation and play from the harddrive, I have been unable to replicate the bug. The CPU does have the F00F bug if that might make some difference. I have attached a backtrace resulting from the attempts to use aplay after the failure has occurred. I may be able to attempt to capture a backtrace at the initial failure if this would be helpful. There are no obvious error messages in my logfiles or printed to the console when the bug occurs. Please let me know how I can assist in squashing this bug. I attempted to use the alsa-info script, but it appears that the database where it uploads the results is broken at the moment. I have pastebinned the results at http://pastebin.ca/1315459 Thanks Stanley Pinchak

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[alsa-devel] alsa-utils compile issues
by Will Wagner 29 Jan '09

29 Jan '09

Hi All, I have configured alsa-lib 1.0.19 to remove the components I do not want for my embedded arm target. I do this with the configure options: --disable-docs \ --disable-aload \ --disable-rawmidi \ --disable-hwdep \ --disable-seq \ --disable-alisp \ --disable-old-symbols \ --disable-python \ --with-alsa-devdir=/dev \ --with-pcm-plugins="copy linear adpcm rate plug dmix softvol" \ --with-ctl-plugins="" \ --with-versioned="no" \ This builds a much reduced in size alsa-lib However when I then try to build alsa-utils this fails because amidi and seq will not compile with the reduced alsa-lib It seems to be there should be configure options in alsa-utils to disable more of its components than just alsamixer and alsatest. I have worked around this by just removing amidi and seq from the SUBDIRS variable in Makefile.in but perhaps there is a better way of doing this? Thanks, Will -- ------------------------------------------------------------------------ Will Wagner will_wagner(a)carallon.com Development Manager Office Tel: +44 (0)20 7371 2032 Carallon Ltd, Studio G20, Shepherds Building, Rockley Rd, London W14 0DA ------------------------------------------------------------------------

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[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 29 Jan '09

29 Jan '09

The following changes since commit 67fcdead3c7424d51e7108c220c9ab1a5e752ed3: Takashi Iwai (1): Merge branch 'topic/snd_card_new-err' into topic/asoc are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Mark Brown (2): ASoC: Fix null string usage with WM8753 DAIs Merge branch 'for-2.6.29' into for-2.6.30 Misael Lopez Cruz (1): ASoC: OMAP: Initialize XCCR and RCCR registers in McBSP DAI driver Peter Ujfalusi (1): ASoC: TWL4030: Add analog loopback support sound/soc/codecs/twl4030.c | 212 ++++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/twl4030.h | 15 +++ sound/soc/codecs/wm8753.c | 9 ++- sound/soc/omap/omap-mcbsp.c | 4 + 4 files changed, 226 insertions(+), 14 deletions(-)

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[alsa-devel] Cannot record from front pluggable mic on 82801H (ICH8 Family) sound card
by Vincent 29 Jan '09

29 Jan '09

Hi, I'm working on a Dell 1525 and I want to plug a mic into the front mic port in order to have a "clean" record. But when it doesn't record anything, all I can have is the noise from the external mic. Is there any specific actions I forgot to do? Here is my alsa-info output : http://www.alsa-project.org/db/?f=a9066fcfe52fa0dd7d9bcd65214855c84592bd25 Thanks, Vincent

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[alsa-devel] ASoC updates for 2.6.29
by Mark Brown 29 Jan '09

29 Jan '09

The following changes since commit 43d50807dbf47a139f83660fec14aebe936b3c80: Peter Ujfalusi (1): ASoC: Add missing comma to SND_SOC_DAPM_SWITCH_E in soc-dapm.h are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.29 Mark Brown (1): ASoC: Fix null string usage with WM8753 DAIs Misael Lopez Cruz (1): ASoC: OMAP: Initialize XCCR and RCCR registers in McBSP DAI driver sound/soc/codecs/wm8753.c | 9 ++++++++- sound/soc/omap/omap-mcbsp.c | 4 ++++ 2 files changed, 12 insertions(+), 1 deletions(-)

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[alsa-devel] [2.6.29-rc3][PATCH 0/1] ASoC: OMAP: Initialize XCCR and RCCR registers in McBSP DAI driver
by Jarkko Nikula 29 Jan '09

29 Jan '09

Hi I'm sending this on behalf of Misael Lopez Cruz with commit id for counterpart patch corrected. This is a fix for 2.6.29. There are some OMAP3 based boards where bootloader is clearing the McBSP XCCR and RCCR register thus causing that DMA is not working with OMAP McBSP DAI driver. As now there is commit 3127f8f8595a064b3f1a1837fea2177902589ac3 in mainline writing those registers, we must initialize them here since otherwise DMA is not working with McBSP DAI driver in any OMAP3 or 2430 based board. -- Jarkko

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[alsa-devel] PCM parameters in file plugin
by Pavel Hofman 29 Jan '09

29 Jan '09

Hello, I am working on a file plugin patch to allow adding PCM parameters (rate, format) to the name of the file created (plus the ability to run an external command via popen, such as upsampling with sox). I got stuck in trying to obtain these parameters in pcm_file.c:_snd_pcm_file_open(). Is it actually possible or the information is not available at the time of opening the file (and the whole patch is thus nonsense)? Thanks a lot for help. Pavel.

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[alsa-devel] [PATCH] sscape: update Kconfig description about SoundScape cards
by Krzysztof Helt 29 Jan '09

29 Jan '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The SoundScape driver handles more cards then described. --- sound/isa/Kconfig | 7 +++++-- 1 files changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index cf6c01d..7d80186 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -378,14 +378,17 @@ config SND_SGALAXY will be called snd-sgalaxy. config SND_SSCAPE - tristate "Ensoniq SoundScape PnP driver" + tristate "Ensoniq SoundScape driver" select SND_HWDEP select SND_MPU401_UART select SND_WSS_LIB help - Say Y here to include support for Ensoniq SoundScape PnP + Say Y here to include support for Ensoniq SoundScape soundcards. + The PCM audio is supported on SoundScape Classic, Elite, PnP + and VIVO cards. The MIDI support is very experimental. + To compile this driver as a module, choose M here: the module will be called snd-sscape. -- 1.5.2.2 ---------------------------------------------------------------------- Promocja w Speak Up. 3 miesiace angielskiego gratis. Sprawdz teraz i wypelnij formularz! >> http://link.interia.pl/f2019

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[alsa-devel] [PATCH] cmi8330: add MPU-401 support
by Krzysztof Helt 29 Jan '09

29 Jan '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Add MPU-401 port support for the chip. Also, update some error messages and description. --- sound/isa/Kconfig | 1 + sound/isa/cmi8330.c | 42 ++++++++++++++++++++++++++++++++++++------ 2 files changed, 37 insertions(+), 6 deletions(-) diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 7d80186..ed798bf 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -95,6 +95,7 @@ config SND_CMI8330 select SND_WSS_LIB select SND_SB16_DSP select SND_OPL3_LIB + select SND_MPU401_UART help Say Y here to include support for soundcards based on the C-Media CMI8330 chip. diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index 1154379..9ca8122 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -31,11 +31,11 @@ * To quickly load the module, * * modprobe -a snd-cmi8330 sbport=0x220 sbirq=5 sbdma8=1 - * sbdma16=5 wssport=0x530 wssirq=11 wssdma=0 + * sbdma16=5 wssport=0x530 wssirq=11 wssdma=0 fmport=0x388 * * This card has two mixers and two PCM devices. I've cheesed it such * that recording and playback can be done through the same device. - * The driver "magically" routes the capturing to the AD1848 codec, + * The driver "magically" routes the capturing to the CMI8330 codec, * and playback to the SB16 codec. This allows for full-duplex mode * to some extent. * The utilities in alsa-utils are aware of both devices, so passing @@ -52,6 +52,7 @@ #include <sound/core.h> #include <sound/wss.h> #include <sound/opl3.h> +#include <sound/mpu401.h> #include <sound/sb.h> #include <sound/initval.h> @@ -81,6 +82,8 @@ static long wssport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; static int wssirq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; static int wssdma[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; static long fmport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static long mpuport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static int mpuirq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for CMI8330 soundcard."); @@ -111,6 +114,10 @@ MODULE_PARM_DESC(wssdma, "DMA for CMI8330 WSS driver."); module_param_array(fmport, long, NULL, 0444); MODULE_PARM_DESC(fmport, "FM port # for CMI8330 driver."); +module_param_array(mpuport, long, NULL, 0444); +MODULE_PARM_DESC(mpuport, "MPU-401 port # for CMI8330 driver."); +module_param_array(mpuirq, int, NULL, 0444); +MODULE_PARM_DESC(mpuirq, "IRQ # for CMI8330 MPU-401 port."); #ifdef CONFIG_PNP static int isa_registered; static int pnp_registered; @@ -153,6 +160,7 @@ struct snd_cmi8330 { #ifdef CONFIG_PNP struct pnp_dev *cap; struct pnp_dev *play; + struct pnp_dev *mpu; #endif struct snd_card *card; struct snd_wss *wss; @@ -169,7 +177,7 @@ struct snd_cmi8330 { #ifdef CONFIG_PNP static struct pnp_card_device_id snd_cmi8330_pnpids[] = { - { .id = "CMI0001", .devs = { { "@@@0001" }, { "@X@0001" } } }, + { .id = "CMI0001", .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" } } }, { .id = "" } }; @@ -329,11 +337,15 @@ static int __devinit snd_cmi8330_pnp(int dev, struct snd_cmi8330 *acard, if (acard->play == NULL) return -EBUSY; + acard->mpu = pnp_request_card_device(card, id->devs[2].id, NULL); + if (acard->play == NULL) + return -EBUSY; + pdev = acard->cap; err = pnp_activate_dev(pdev); if (err < 0) { - snd_printk(KERN_ERR "CMI8330/C3D (AD1848) PnP configure failure\n"); + snd_printk(KERN_ERR "CMI8330/C3D PnP configure failure\n"); return -EBUSY; } wssport[dev] = pnp_port_start(pdev, 0); @@ -354,6 +366,17 @@ static int __devinit snd_cmi8330_pnp(int dev, struct snd_cmi8330 *acard, sbdma16[dev] = pnp_dma(pdev, 1); sbirq[dev] = pnp_irq(pdev, 0); + /* allocate MPU-401 resources */ + pdev = acard->mpu; + + err = pnp_activate_dev(pdev); + if (err < 0) { + snd_printk(KERN_ERR + "CMI8330/C3D (MPU-401) PnP configure failure\n"); + return -EBUSY; + } + mpuport[dev] = pnp_port_start(pdev, 0); + mpuirq[dev] = pnp_irq(pdev, 0); return 0; } #endif @@ -502,11 +525,11 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev) wssdma[dev], -1, WSS_HW_DETECT, 0, &acard->wss); if (err < 0) { - snd_printk(KERN_ERR PFX "(AD1848) device busy??\n"); + snd_printk(KERN_ERR PFX "(CMI8330) device busy??\n"); return err; } if (acard->wss->hardware != WSS_HW_CMI8330) { - snd_printk(KERN_ERR PFX "(AD1848) not found during probe\n"); + snd_printk(KERN_ERR PFX "(CMI8330) not found during probe\n"); return -ENODEV; } @@ -552,6 +575,13 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev) } } + if (mpuport[dev] != SNDRV_AUTO_PORT) { + if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, + mpuport[dev], 0, mpuirq[dev], + IRQF_DISABLED, NULL) < 0) + printk(KERN_ERR PFX "no MPU-401 device at 0x%lx.\n", + mpuport[dev]); + } strcpy(card->driver, "CMI8330/C3D"); strcpy(card->shortname, "C-Media CMI8330/C3D"); -- 1.5.2.2 ---------------------------------------------------------------------- Obdaruj swoja Walentynke ... lub siebie! Wygraj nagrody! Sprawdz >> http://link.interia.pl/f203a

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[alsa-devel] [PATCH] gus: update debug messages
by Krzysztof Helt 29 Jan '09

29 Jan '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Convert some of them to snd_printdd() and update arguments to make them compilable. --- sound/isa/gus/gus_dma.c | 24 +++++++++++++++--------- 1 files changed, 15 insertions(+), 9 deletions(-) diff --git a/sound/isa/gus/gus_dma.c b/sound/isa/gus/gus_dma.c index f45f611..cf8cd3c 100644 --- a/sound/isa/gus/gus_dma.c +++ b/sound/isa/gus/gus_dma.c @@ -45,7 +45,8 @@ static void snd_gf1_dma_program(struct snd_gus_card * gus, unsigned char dma_cmd; unsigned int address_high; - // snd_printk("dma_transfer: addr=0x%x, buf=0x%lx, count=0x%x\n", addr, (long) buf, count); + snd_printdd("dma_transfer: addr=0x%x, buf=0x%lx, count=0x%x\n", + addr, buf_addr, count); if (gus->gf1.dma1 > 3) { if (gus->gf1.enh_mode) { @@ -142,7 +143,9 @@ static void snd_gf1_dma_interrupt(struct snd_gus_card * gus) snd_gf1_dma_program(gus, block->addr, block->buf_addr, block->count, (unsigned short) block->cmd); kfree(block); #if 0 - printk("program dma (IRQ) - addr = 0x%x, buffer = 0x%lx, count = 0x%x, cmd = 0x%x\n", addr, (long) buffer, count, cmd); + snd_printd(KERN_DEBUG "program dma (IRQ) - " + "addr = 0x%x, buffer = 0x%lx, count = 0x%x, cmd = 0x%x\n", + block->addr, block->buf_addr, block->count, block->cmd); #endif } @@ -203,13 +206,16 @@ int snd_gf1_dma_transfer_block(struct snd_gus_card * gus, } *block = *__block; block->next = NULL; -#if 0 - printk("addr = 0x%x, buffer = 0x%lx, count = 0x%x, cmd = 0x%x\n", block->addr, (long) block->buffer, block->count, block->cmd); -#endif -#if 0 - printk("gus->gf1.dma_data_pcm_last = 0x%lx\n", (long)gus->gf1.dma_data_pcm_last); - printk("gus->gf1.dma_data_pcm = 0x%lx\n", (long)gus->gf1.dma_data_pcm); -#endif + + snd_printdd("addr = 0x%x, buffer = 0x%lx, count = 0x%x, cmd = 0x%x\n", + block->addr, (long) block->buffer, block->count, + block->cmd); + + snd_printdd("gus->gf1.dma_data_pcm_last = 0x%lx\n", + (long)gus->gf1.dma_data_pcm_last); + snd_printdd("gus->gf1.dma_data_pcm = 0x%lx\n", + (long)gus->gf1.dma_data_pcm); + spin_lock_irqsave(&gus->dma_lock, flags); if (synth) { if (gus->gf1.dma_data_synth_last) { -- 1.5.2.2 ---------------------------------------------------------------------- Speak Up. Angielski szybko i skutecznie. 3 miesiace nauki gratis. Sprawdz. >> http://link.interia.pl/f2019

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[alsa-devel] (no subject)
by Philippe CARRIERE 29 Jan '09

29 Jan '09

Hi, following is a report of some tests of alsa.1.0.18rc3 in view of support of Tascam US-122L, which might be of interest for you. As a preliminary information, note that I patched my own Fedora kernel and alsa-plugins with Karsten Wiese'patches a few month ago and use regularly Tascam US-122L under Linux which works very fine; I also provide (experimental) packages for Fedora 9 from june 2008 on my web page. Test 1: succeeded - replacement of my own alsa-plugins-usb_stream-1.0.16-4.us122lv05.fc9 by alsa-plugins-usbstream-1.0.18-1.rc3.fc10 on Fedora 9 (using my own patched kernel 2.6.26 including the snd-usb-us122l module). Everything works. Test 2: succeeded - replacement of previous alsa-1.0.16 by alsa.1.0.18-1.rc3 (Fedora packages; thus alsa-driver-1.0.18rc3 not included). Everything works, including Tascam US-122L when using my patched kernel (which includes the snd-usb-us122l module). Test 3: failed - installation of alsa-driver-1.0.18rc3 (succeeded) on Fedora 9 (as weel, I tested with new Fedora 10 beta which includes kernel 2.6.27.rc3); plug in the US-122L leads to error messages from snd-usb-us122l: # cat /var/log/messages ... Oct 11 09:21:51 pollux kernel: usb 2-6: new high speed USB device using ehci_hcd and address 3 Oct 11 09:21:51 pollux kernel: usb 2-6: config 1 interface 1 altsetting 1 bulk endpoint 0x83 has invalid maxpacket 9 Oct 11 09:21:51 pollux kernel: usb 2-6: configuration #1 chosen from 1 choice Oct 11 09:21:51 pollux kernel: usb 2-6: New USB device found, idVendor=0644, idProduct=800e Oct 11 09:21:51 pollux kernel: usb 2-6: New USB device strings: Mfr=1, Product=2, SerialNumber=3 Oct 11 09:21:51 pollux kernel: usb 2-6: Product: US-122L Oct 11 09:21:51 pollux kernel: usb 2-6: Manufacturer: TASCAM Oct 11 09:21:51 pollux kernel: usb 2-6: SerialNumber: no serial number Oct 11 09:21:51 pollux kernel: ALSA /home/Test/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:618: 0 Oct 11 09:21:55 pollux kernel: ALSA /home/Test/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/us122l.c:329: us122l_start error -14 Oct 11 09:21:55 pollux kernel: snd-usb-us122l: probe of 2-6:1.1 failed with error -5 Oct 11 09:21:55 pollux kernel: usbcore: registered new interface driver snd-usb-us122l and # cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xfebfc000 irq 21 US-122L finally not being recognized. Test 4: succeded - same than test 3 for Fedora 9, but using a patched ehci-hcd module (the patch adding some features from the original one by Karsten Wiese not - yet ? - included in kernel 2.6.26). Everything works including of course the US-122L. Question: Thus, a patched ehci-hcd module is required. Is the patch scheduled to appear in kernel 2.6.27 or would it be useful to work for providing a kind of kmod ? Regards Phil. -- Philippe Carriere <philippe-f.carriere(a)wanadoo.fr>

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[RESEND][PATCH] USB: Prevent EHCI ITDs reusage while frame is active: tested by: Philippe Carriere.
by Philippe Carriere 29 Jan '09

29 Jan '09

Hi, I successfully tested the refreshed patch of Karsten using Fedora rawhide packages. Following are some informations on the test: original kernel: 2.6.29-0.53.rc2.git1.fc11.x86_64 patched kernel: Linux 2.6.29-0.53.rc2.git1.ehcius122l.fc11.x86_64 #1 SMP Wed Jan 28 22:48:01 CET 2009 x86_64 x86_64 x86_64 GNU/Linux alsa: alsa-lib-1.0.19-1.fc11, alsa-plugins-usbstream-1.0.18 jack: jack-audio-connection-kit-0.116.1-3.fc11 qjackctl: qjackctl-0.3.3-1.fc10 audacious: audacious-1.5.1-5.fc11 after plugin the Tascam US-122L: [root@pollux Test]# tail /var/log/messages Jan 29 08:05:53 pollux kernel: usb 2-6: new high speed USB device using ehci_hcd and address 4 Jan 29 08:05:53 pollux kernel: usb 2-6: config 1 interface 1 altsetting 1 bulk endpoint 0x83 has invalid maxpacket 9 Jan 29 08:05:53 pollux kernel: usb 2-6: New USB device found, idVendor=0644, idProduct=800e Jan 29 08:05:53 pollux kernel: usb 2-6: New USB device strings: Mfr=1, Product=2, SerialNumber=3 Jan 29 08:05:53 pollux kernel: usb 2-6: Product: US-122L Jan 29 08:05:53 pollux kernel: usb 2-6: Manufacturer: TASCAM Jan 29 08:05:53 pollux kernel: usb 2-6: SerialNumber: no serial number Jan 29 08:05:53 pollux kernel: usb 2-6: configuration #1 chosen from 1 choice Jan 29 08:06:08 pollux kernel: ALSA sound/usb/usx2y/usb_stream.c:457: underrun, status=4294967278 Jan 29 08:06:08 pollux kernel: ALSA sound/usb/usx2y/usb_stream.c:457: underrun, status=4294967278 [root@pollux Test]# cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xfebfc000 irq 21 1 [US122L ]: USB US-122L - TASCAM US-122L TASCAM US-122L (644:800e if 0 at 002/004) and jack messages were 08:50:00.226 /usr/bin/jackd -R -P89 -dalsa -dusb_stream:1 -r48000 -p512 -n2 jackd 0.116.1 Copyright 2001-2005 Paul Davis and others. jackd comes with ABSOLUTELY NO WARRANTY This is free software, and you are welcome to redistribute it under certain conditions; see the file COPYING for details JACK compiled with System V SHM support. 08:50:00.251 JACK was started with PID=6417. loading driver .. apparent rate = 48000 creating alsa driver ... usb_stream:1|usb_stream:1|512|2|48000|0|0| nomon|swmeter|-|32bit ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL usb_stream:1 control open "usb_stream:1" (No such file or directory) configuring for 48000Hz, period = 512 frames (10.7 ms), buffer = 2 periods ALSA: final selected sample format for capture: 24bit little-endian ALSA: use 2 periods for capture ALSA: final selected sample format for playback: 24bit little-endian ALSA: use 2 periods for playback 08:50:02.397 Server configuration saved to "/home/Test/.jackdrc". 08:50:02.398 Statistics reset. 08:50:02.547 Client activated. 08:50:02.550 JACK connection change. 08:50:02.558 JACK connection graph change. Tested by .mp3 file playback using audacious (with plugin jack). Additional notes: the previous version of the patch is routinely used by myself, at least some Fedora users using my packages at http://pagesperso-orange.fr/La-page-Web-of-Phil/Tascam_US-122L/Telechargeme… , and under Debian and Ubuntu (Federico Briata web site at http://wiki.briata.org/doku.php?id=testing_us122l_under_linux ) under 2.6.26 and 2.6.27 kernel. Hope it will be soon integrated to 2.6.29 devel. Regards Phil.

1 0

[alsa-devel] Misusing snd_pcm_avail_update()
by Lennart Poettering 29 Jan '09

29 Jan '09

Heya! Currently in the 'glitch-free' logic of PulseAudio I use snd_pcm_avail_update() to estimate how I need to program my system timers for the next wake-up for the next buffer fill-up. For that I assume that the current fill level of the hardware buffer is the hardware buffer size minus what s_p_a_u() returns. I then convert that fill level from sample units to time units, and fix it up by the deviation of the sound card time from the system time. Finally I substract some extra margin just to make sure. This I assumed would tell me how much time will pass until an underrun happens if I don't write anything. Mostly this logic works fine. But on some setups and cases it doesn't. ALSA will signal an underrun much much earlier than what I estimated like this. I am now wondering why? One possibility of course is that s_p_a_u() is not reliable, due to driver issues (there were problems in the HDA driver about this, right?). Also, s_p_a_u() might simply lag behind quite a bit, or -- what I think is most likely -- because samples are popped in larger blocks form the hw playback buffer we reach the underrun much earlier than expected. I do acknowledge that the way i use s_p_a_u() is probably a misuse of the API. I make asssumptions I probably shouldn't make. Now, considering all this I'd like to ask for a new API function that tells me how much time I really have before the next underrun. It probably should just return a value in sample units, leaving it for the application to deal with system/sound card clock deviations. Any opinions on this? Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] Strange Config: Fedora 8 running on Amazon EC2
by Patrick Lightbody 29 Jan '09

29 Jan '09

I have a strange request (which is why I'm sending directly to the dev list). I'm trying to run Flash video (v9 or v10) on Firefox 3 on a Fedora 8 image running on Amazon EC2. I don't actually want/need sound (it's of course technically not possible, since EC2 doesn't provide a sound card). Video playback (when watched through VNC) goes fine for a couple minutes, but then the browser and flash lock up entirely. I've looked for ways to tell Firefox and/or Flash to not send audio out, but I've come up empty so far. So now I'm hoping that fixing this error and putting in perhaps some sort of "no op" device might help out. I'm not positive the error is related to ALSA, though the fact that there appear to be a ton of the following logs just before the lockup is suspicious: ALSA lib confmisc.c:768:(parse_card) cannot find card '0' ALSA lib conf.c:3513:(_snd_config_evaluate) function snd_func_card_driver returned error: No such file or directory ALSA lib confmisc.c:392:(snd_func_concat) error evaluating strings ALSA lib conf.c:3513:(_snd_config_evaluate) function snd_func_concat returned error: No such file or directory ALSA lib confmisc.c:1251:(snd_func_refer) error evaluating name ALSA lib conf.c:3513:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory ALSA lib conf.c:3985:(snd_config_expand) Evaluate error: No such file or directory ALSA lib pcm.c:2144:(snd_pcm_open_noupdate) Unknown PCM default ALSA lib pcm_hw.c:1240:(_snd_pcm_hw_open) Invalid value for card (these lines repeat over and over again until the crash happens, then it stops promptly) As you can see by the alsa-info output at the following URL, I don't have any running snd modules: http://www.alsa-project.org/db/?f=6b2431d0564b3638cdb798a6d1a21eb73982588c Similarly, I don't have any sound-related devices installed (ie: /dev/audio, etc). My hope in emailing this list is that there may be a solution that causes ALSA to no-op the requests that Firefox/Flash send it's way, potentially solving the lockup I'm experiencing. I would really appreciate any ideas that could be shared. Thank you very much! Patrick

2 2

[alsa-devel] ASoC : underrun messages
by Naveen Krishna Ch 28 Jan '09

28 Jan '09

Hello, I am experiencing some audio issue While playing using aplay some times i see the following messages. (Sorry if the messages are too long) [root@OMAP3 audio_samples]#aplay sample_96K.wav Playing WAVE 'sample_96K.wav' : Signed 16 bit Little Endian, Rate 96000 Hz, Stereo underrun!!! (at least 1307.251 ms long) underrun!!! (at least 554.656 ms long) underrun!!! (at least 988.861 ms long) underrun!!! (at least 554.443 ms long) underrun!!! (at least 987.670 ms long) underrun!!! (at least 995.484 ms long) ^CAborted by signal Interrupt... [root@OMAP3 audio_samples]#aplay sample_48K.wav Playing WAVE 'sample_48K.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo underrun!!! (at least 69.306 ms long) underrun!!! (at least 130.981 ms long) underrun!!! (at least 205.750 ms long) underrun!!! (at least 178.100 ms long) underrun!!! (at least 146.057 ms long) ^CAborted by signal Interrupt... [root@OMAP3 audio_samples]#aplay sample_8K.wav Playing WAVE 'sample_8K.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Stereo ^C^[[AAborted by signal Interrupt... [root@OMAP3 audio_samples]#aplay sample_22K.wav Playing WAVE 'sample_22K.wav' : Signed 16 bit Little Endian, Rate 22050 Hz, Stereo ^CAborted by signal Interrupt... [root@OMAP3 audio_samples]#aplay sample_44K.wav Playing WAVE 'sample_44K.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo underrun!!! (at least 1602.264 ms long) underrun!!! (at least 873.078 ms long) underrun!!! (at least 1693.268 ms long) underrun!!! (at least 870.483 ms long) underrun!!! (at least 540.466 ms long) underrun!!! (at least 1122.711 ms long) I am using OMAP3430 + twl4030 ASoC audio codec drivers. I understand that the prints are coming from alsa-lib aplay.c file As a result of the client not reading the buffers fast enough. These are not conditionally reproducible, i see them uncertainly. I saw the same isue with 2.6.26 some one suggested me to find if there is any bug in the asoc layer. Now i am using 2.6.28 kernel from kernel.org but still got the same Can any one suggest me a way to find a fix for this.. -- Cheers (: Naveen Krishna Ch :)

2 2

[alsa-devel] alsa-lib failing at runtime
by Will Wagner 28 Jan '09

28 Jan '09

Hi All, I have found a problem with alsa-lib 1.0.19 running on an embedded arm board. It all compiles ok but aplay fails to play a .wav file and the kernel reports this error: ALSA sound/core/pcm_native.c:2587: unknown ioctl = 0x80184132 Decoding this ioctl it is SNDRV_PCM_IOCTL_CHANNEL_INFO defined in asound.h. The reason it is not working is that the kernel and alsa-lib think that the sndrv_pcm_channel_info struct is different sizes. The definition in asound.h is: struct sndrv_pcm_channel_info { unsigned int channel; off_t offset; /* mmap offset */ unsigned int first; /* offset to first sample in bits */ unsigned int step; /* samples distance in bits */ }; The problem is that when compiling linux it thinks off_t is 4 bytes but when compiling alsa-lib it thought it was 8 bytes. After discussion on the buildroot mailing list (see http://lists.busybox.net/pipermail/buildroot/2009-January/025458.html) it seems like for the kernel off_t is always 4 bytes but for user space it can be either 4 or 8 bytes depending on whether large file support is enabled. It appears that the fix is that the type of offset should be changed to long. Hope this all makes sense. Thanks, Will. -- ------------------------------------------------------------------------ Will Wagner will_wagner(a)carallon.com Development Manager Office Tel: +44 (0)20 7371 2032 Carallon Ltd, Studio G20, Shepherds Building, Rockley Rd, London W14 0DA ------------------------------------------------------------------------

2 1

[alsa-devel] ASoC updates
by Mark Brown 28 Jan '09

28 Jan '09

The following changes since commit 070504ade7a95a0f4395673717f3bb7d41793ca8: Mark Brown (1): ASoC: Fix L3 bus handling in Kconfig are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Mark Brown (1): ASoC: Push the codec runtime storage into the card structure Peter Ujfalusi (5): ASoC: TWL4030: Syncronize the reg_cache for ANAMICL after the offset cancelation ASoC: TWL4030: Code clean up for codec power up and down ASoC: TWL4030: Enable Headset Left anti-pop/bias ramp only if the Headset Left is in use ASoC: TWL4030: Physical ADC and amplifier power switch change ASoC: TWL4030: Move the twl4030_power_up and _power_down function Timur Tabi (1): ASoC: improve I2C initialization code in CS4270 driver include/sound/soc.h | 3 +- sound/soc/codecs/ac97.c | 20 ++-- sound/soc/codecs/ad1980.c | 12 +- sound/soc/codecs/ad73311.c | 8 +- sound/soc/codecs/ak4535.c | 14 +- sound/soc/codecs/cs4270.c | 254 ++++++++++++++++++---------------------- sound/soc/codecs/pcm3008.c | 12 +- sound/soc/codecs/ssm2602.c | 20 ++-- sound/soc/codecs/tlv320aic23.c | 18 ++-- sound/soc/codecs/tlv320aic26.c | 4 +- sound/soc/codecs/tlv320aic3x.c | 14 +- sound/soc/codecs/twl4030.c | 224 +++++++++++++++++++----------------- sound/soc/codecs/uda134x.c | 18 ++-- sound/soc/codecs/uda1380.c | 18 ++-- sound/soc/codecs/wm8350.c | 10 +- sound/soc/codecs/wm8510.c | 16 ++-- sound/soc/codecs/wm8580.c | 10 +- sound/soc/codecs/wm8728.c | 16 ++-- sound/soc/codecs/wm8731.c | 20 ++-- sound/soc/codecs/wm8750.c | 16 ++-- sound/soc/codecs/wm8753.c | 18 ++-- sound/soc/codecs/wm8900.c | 8 +- sound/soc/codecs/wm8903.c | 18 ++-- sound/soc/codecs/wm8971.c | 14 +- sound/soc/codecs/wm8990.c | 14 +- sound/soc/codecs/wm9705.c | 15 ++- sound/soc/codecs/wm9712.c | 21 ++-- sound/soc/codecs/wm9713.c | 17 ++-- sound/soc/soc-core.c | 37 +++--- sound/soc/soc-dapm.c | 6 +- sound/soc/soc-jack.c | 4 +- 31 files changed, 443 insertions(+), 456 deletions(-)

2 1

[alsa-devel] [PATCH] ALSA: hda - add another MacBook Pro 4, 1 subsystem ID
by Luke Yelavich 28 Jan '09

28 Jan '09

Add another MacBook Pro 4,1 SSID (106b:3800). It seems that latter revisions, (at least mine), have different IDs to earlier revisions. Signed-off-by: Luke Yelavich <themuso(a)ubuntu.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 863ab95..aef543a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7036,6 +7036,7 @@ static int patch_alc882(struct hda_codec *codec) case 0x106b00a4: /* MacbookPro4,1 */ case 0x106b2c00: /* Macbook Pro rev3 */ case 0x106b3600: /* Macbook 3.1 */ + case 0x106b3800: /* MacbookPro4,1 - latter revision */ board_config = ALC885_MBP3; break; default: -- 1.5.6.3

2 1

[alsa-devel] gusclassic alsa configuration
by Krzysztof Helt 28 Jan '09

28 Jan '09

Hi, I have given another go to fix Gravis Ultrasound configuration for ALSA and I have found a working configuration. The Gravis Classic card uses non-mmap and non-interleaved sound format so aplay fails with standard configuration. Even worse, no default pcm device is defined so someone who wants just install Linux on old computer won't get any sound. I have modified the GUS.conf from the /usr/share/alsa/cards and the GUS card now works with alsa and oss. My changes to the GUS.conf are below (as a diff). Does anybody know how to upload MIDI instruments into the GUS card under ALSA? (like with awesfx for SB AWE) Regards, Krzysztof --- /usr/src/alsa/alsa-lib-1.0.18/src/conf/cards/GUS.conf 2008-10-29 13:42:10.000000000 +0100 +++ GUS.conf 2009-01-21 20:42:42.675215220 +0100 @@ -19,10 +19,12 @@ } # -# It's a temporary solution. +# First route two channels to two other channels +# to drop mmap requirement. Then, use the mmap_emul +# plugin to allow using noninterleaved channels. # -!pcm.hw { +!pcm.default { @args [ CARD DEV SUBDEV ] @args.CARD { type string @@ -58,9 +60,16 @@ name defaults.pcm.subdevice } } - type hw - card $CARD - device $DEV - subdevice $SUBDEV - mmap_emulation on + type route + slave.pcm { + type mmap_emul + slave.pcm { + type hw + card $CARD + device $DEV + subdevice $SUBDEV + } + } + ttable.0.0 1 + ttable.1.1 1 } ---------------------------------------------------------------------- Speak Up. Angielski szybko i skutecznie. 3 miesiace nauki gratis. Sprawdz. >> http://link.interia.pl/f2019

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Re: [alsa-devel] [PATCH] Turtle Beach Multisound Classic/Pinnacle driver (4th rev)
by Takashi Iwai 28 Jan '09

28 Jan '09

At Sat, 24 Jan 2009 13:35:28 +0100, Krzysztof Helt wrote: > > From: Krzysztof Helt <krzysztof.h1(a)wp.pl> > > This is driver for Turtle Beach Multisound cards: > Classic, Fiji and Pinnacle. > > Tested pcm playback and recording and MIDI playback > on Multisound Pinnacle. > > Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> > > --- > > All issues pointed by Takashi are resolved. I moved > more code to the common part and made a common > library. Thanks, this looks pretty good. But, one thing I forgot to ask... > +static int __devinit snd_msnd_isa_probe(struct device *pdev, unsigned int idx) > +{ > + int err; > + struct snd_card *card; > + struct snd_msnd *chip; > + > + if (isapnp[idx] || cfg[idx] == SNDRV_AUTO_PORT) { > + printk(KERN_INFO LOGNAME ": Assuming PnP mode\n"); > + return -ENODEV; > + } > + > + card = snd_card_new(index[idx], id[idx], THIS_MODULE, > + sizeof(struct snd_msnd)); > + if (!card) > + return -ENOMEM; Could you use the new API snd_card_create() introduced recently? > +#ifdef CONFIG_PNP > +static int __devinit snd_msnd_pnp_detect(struct pnp_card_link *pcard, > + const struct pnp_card_device_id *pid) > +{ ... > + /* > + * Create a new ALSA sound card entry, in anticipation > + * of detecting our hardware ... > + */ > + card = snd_card_new(index[idx], id[idx], THIS_MODULE, > + sizeof(struct snd_msnd)); > + if (!card) > + return -ENOMEM; Ditto. thanks, Takashi

2 2

[alsa-devel] [PATCH 0/6] ASoC: TWL4030: Cleanups and analog loopback support
by Peter Ujfalusi 27 Jan '09

27 Jan '09

Hello, The following series - at the end - will add analog loopback/bypass support for the twl4030 codec. The main changes: [1] ANAMICL register self cleaning bit handling [2] Simplified power up/down function [3] Move the Headset anti-pop/bias ramp enabling to be happend only if the headset is actually in use [4] Change the place for the DAC power switch [5] Preparation for the loopback implementation [6] Analog loopback/bypass implementation You can find a bit more detailed notes in the actuall patches. In short: if none of the loopbacks are enabled, the codec behaves in the same way as it used to. If the loopback is enabled the codec will be kept powered as long as the loopback is enabled. P�ter --- Peter Ujfalusi (6): ASoC: TWL4030: Syncronize the reg_cache for ANAMICL after the offset cancelation ASoC: TWL4030: Code clean up for codec power up and down ASoC: TWL4030: Enable Headset Left anti-pop/bias ramp only if the Headset Left is in use ASoC: TWL4030: Physical ADC and amplifier power switch change ASoC: TWL4030: Move the twl4030_power_up and _power_down function ASoC: TWL4030: Add analog loopback support sound/soc/codecs/twl4030.c | 340 ++++++++++++++++++++++++++++++-------------- 1 files changed, 232 insertions(+), 108 deletions(-)

2 12

[alsa-devel] [PATCH] Detailed messages in usbaudio.c
by LCID Fire 27 Jan '09

27 Jan '09

Attached are 3 patches which make the error messages in usbaudio.c more useful and also some code optimizations. commit 191cf816a5a618a84cc3627c5f9f96355f5d22c4 Author: Andreas Bergmeier <lcid-fire(a)gmx.net> Date: Sat Jan 24 12:17:01 2009 +0100 Print device number which did not support low speed audio streaming Signed-off-by: Andreas Bergmeier <lcid-fire(a)gmx.net> diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 8b8eb9e..2cfb8a3 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2929,7 +2929,7 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) continue; } if (snd_usb_get_speed(dev) == USB_SPEED_LOW) { - snd_printk(KERN_ERR "low speed audio streaming not supported\n"); + snd_printk(KERN_ERR "%d: low speed audio streaming not supported\n", dev->devnum); continue; } if (! parse_audio_endpoints(chip, j)) { commit c6d62cbaf6d36b252768ef109b8d85a036e1d451 Author: Andreas Bergmeier <lcid-fire(a)gmx.net> Date: Sat Jan 24 12:16:16 2009 +0100 Replaced 'return NULL' lines by break in snd_usb_find_desc so there is only one default error. Signed-off-by: Andreas Bergmeier <lcid-fire(a)gmx.net> diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index d35b51c..8b8eb9e 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2054,10 +2054,10 @@ void *snd_usb_find_desc(void *descstart, int desclen, void *after, u8 dtype) end = p + desclen; for (; p < end;) { if (p[0] < 2) - return NULL; + break; next = p + p[0]; if (next > end) - return NULL; + break; if (p[1] == dtype && (!after || (void *)p > after)) { return p; } commit cd0180399253b17ece4fc1177457b21b8b70fe8f Author: Andreas Bergmeier <lcid-fire(a)gmx.net> Date: Sat Jan 24 12:22:13 2009 +0100 Add device and control number so it's apparent for which control this error is showing up. Signed-off-by: Andreas Bergmeier <lcid-fire(a)gmx.net> diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 037435d..e578a5c 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2867,12 +2867,12 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) /* find audiocontrol interface */ host_iface = &usb_ifnum_to_if(dev, ctrlif)->altsetting[0]; if (!(p1 = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, NULL, HEADER))) { - snd_printk(KERN_ERR "cannot find HEADER\n"); + snd_printk(KERN_ERR "%d:%u: cannot find HEADER\n", dev->devnum, ctrlif); return -EINVAL; } if (! p1[7] || p1[0] < 8 + p1[7]) { - snd_printk(KERN_ERR "invalid HEADER\n"); + snd_printk(KERN_ERR "%d:%u: invalid HEADER\n", dev->devnum, ctrlif); return -EINVAL; }

2 3

Re: [alsa-devel] handle an external PWM amplifier via alsa interface
by Mark Brown 27 Jan '09

27 Jan '09

On Tue, Jan 27, 2009 at 11:15:48AM +0100, Matteo Canella wrote: > I'm trying to implement an alsa interface for the audio section of my board. > I have an mpc5200 that control via i2s a pcm1774 stereo DAC and I also have > an amplifier between the DAC and the speakers. > At the moment I have the pcm1774 driver properly working, but I have a few > ideas on how to integrate the control of the external amplifier in the alsa > structure. I need to set the amplifier level with a PWM input to it. > Now I'm trying to use "non dapm" controls, adding the PWM settings into this > structure... > What do you guys think about this? > Am I completely wrong or could this be a good way? Using a non-DAPM control to set gain sounds sensible. Depending on how exactly this works you may want to also have a DAPM control so you can power the amplifier off when the path isn't in use (though your application may not be sufficiently sensitive to power consumption to make it worth bothering or the hardware may not let you do anything useful).

1 0

[alsa-devel] Help with Behringer UCG102 Quirks
by Keith A. Milner 27 Jan '09

27 Jan '09

Hi, I'm new to alsa development work so I would appreciate some advice. I'm trying to add specific support for the Behringer UCG102 Guitar Link USB interface. This is a simple device based on a TI codec. The aim is twofold: a) To provide a more descriptive name, as it currently describes itself as the generic TI device This was easy... I've included the following in usbquirks.h: /* Behringer UCG102 Guitar Link */ { USB_DEVICE(0x08bb, 0x2902), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Behringer", .product_name = "Guitar Link UCG102", .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, } }, b) To include more specific support for the characteristics of the UCG102, such as sample rate and so on (if required). I suspect something is required in this respect because when you connect it, dmesg reports "cannot get freq at ep 0x84". I also get a lot of xruns with this device. I suspect it's not reporting itself properly to ALSA and isn't getting set up with the right settings. Of course, I may be completely wrong. I've attached the lsusb output for this. Can anyone help me decode this and help me understand what settings I need to configure? Cheers, -- Keith A. Milner

1 1

[alsa-devel] handle an external PWM amplifier via alsa interface
by Matteo Canella 27 Jan '09

27 Jan '09

Hi, I'm a newbie in ALSA development. I'm trying to implement an alsa interface for the audio section of my board. I have an mpc5200 that control via i2s a pcm1774 stereo DAC and I also have an amplifier between the DAC and the speakers. At the moment I have the pcm1774 driver properly working, but I have a few ideas on how to integrate the control of the external amplifier in the alsa structure. I need to set the amplifier level with a PWM input to it. Now I'm trying to use "non dapm" controls, adding the PWM settings into this structure... What do you guys think about this? Am I completely wrong or could this be a good way? Thanks a lot Ing. Matteo Canella Firmware Engineer Digitek - MTA SpA Via L. Romagnoli, 24 41033 Concordia s.S. (MO) - Italy T. +39 0535 57732 F. +39 0535 54550 email: matteo.canella(a)mta.it

1 0

[alsa-devel] CMI8788/AV200/Oxygen Volume Controls in alsamixer (oxygen_mixer.c:dac_volume_get())
by John L. Utz III 27 Jan '09

27 Jan '09

Hi Clemens and all; ( To those following along at home, kindly open alsa-driver-1.0.18a/alsa-kernel/pci/oxygen/oxygen_mixer.c and look at dac_volume_get() and dac_volume_put() ) It appears to me that these functions are attempting to work on all of the channels at once instead of just choosing Master, Front, Rear, Surround, or what have you: static int dac_volume_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) { struct oxygen *chip = ctl->private_data; unsigned int i; mutex_lock(&chip->mutex); for (i = 0; i < chip->model.dac_channels; ++i) value->value.integer.value[i] = chip->dac_volume[i]; mutex_unlock(&chip->mutex); return 0; } This is then surfaced to userland thusly in the controls array: static const struct snd_kcontrol_new controls[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Volume", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, .info = dac_volume_info, .get = dac_volume_get, .put = dac_volume_put, }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", .info = snd_ctl_boolean_mono_info, .get = dac_mute_get, .put = dac_mute_put, }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Stereo Upmixing", .info = upmix_info, .get = upmix_get, .put = upmix_put, }, { blah }, { blah }, { blah } }; this leads to a rather awkward result in alsamixer where there are 5 volume controls called "Master": < Master > Master Master Master Master IMHO, it would be more desirable for them to say: < Master > Front C/LFE Surround Rear Surr Our app would certainly be much happier :-) So, i *think* that i can fix this thusly: static const struct snd_kcontrol_new controls[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Volume", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, .info = dac_volume_info, .get = dac_volume_get, .put = dac_volume_put, .private_value = 0 }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", .info = snd_ctl_boolean_mono_info, .get = dac_mute_get, .put = dac_mute_put, .private_value = 0 }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Front Playback Volume", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, .info = dac_volume_info, .get = dac_volume_get, .put = dac_volume_put, .private_value = 1 }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Front Playback Switch", .info = snd_ctl_boolean_mono_info, .get = dac_mute_get, .put = dac_mute_put, .private_value = 1 }, and alter dac_volume_*() to not use a for() loop and just work off the private_value. anybody have any opinions? i'll probably have tried this already before anybody responds but it seemed reasonable to surface the idea. tnx! johnu

4 12

[alsa-devel] ICE1712 softdevices fixes
by Diego E. 'Flameeyes' Pettenò 26 Jan '09

26 Jan '09

The following patches fixes the default device as they appear for ICE1712-based cards. Without this cards, 'aplay -L' will show a series of devices like surround40 and surround71 that are not meaningful for this card, and a front device that would not work. Diego E. 'Flameeyes' Pettenò (2): Remove definition for surround sound devices on ICE1712. Make the front:idx device work for ICE1712.

5 13

[alsa-devel] [PATCH] hda: 92hd71xxx disable unmute support for codecs that don't have input amps.
by Matthew Ranostay 26 Jan '09

26 Jan '09

Some revisions of the 92hd71xxx codec families don't have input amps on ports 0xa, 0xd and 0xf, so probe the widget caps on port 0xa and check for support, if found run snd_hda_sequence_write_cache() on the stac92hd71xxx_unmute_core_init verb list. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- sound/pci/hda/patch_sigmatel.c | 20 ++++++++++++-------- 1 files changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 80a4c28..ffef091 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -858,26 +858,25 @@ static struct hda_verb stac92hd83xxx_core_init[] = { static struct hda_verb stac92hd71bxx_core_init[] = { /* set master volume and direct control */ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ - { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {} }; -#define HD_DISABLE_PORTF 2 +#define HD_DISABLE_PORTF 1 static struct hda_verb stac92hd71bxx_analog_core_init[] = { /* start of config #1 */ /* connect port 0f to audio mixer */ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, - /* unmute right and left channels for node 0x0f */ - { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* start of config #2 */ /* set master volume and direct control */ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* unmute right and left channels for nodes 0x0a, 0xd */ + {} +}; + +static struct hda_verb stac92hd71bxx_unmute_core_init[] = { + /* unmute right and left channels for nodes 0x0f, 0xa, 0x0d */ + { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {} @@ -4942,6 +4941,7 @@ static struct hda_input_mux stac92hd71bxx_dmux = { static int patch_stac92hd71bxx(struct hda_codec *codec) { struct sigmatel_spec *spec; + struct hda_verb *unmute_init = stac92hd71bxx_unmute_core_init; int err = 0; spec = kzalloc(sizeof(*spec), GFP_KERNEL); @@ -5015,6 +5015,7 @@ again: /* disable VSW */ spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF]; + unmute_init++; stac_change_pin_config(codec, 0xf, 0x40f000f0); break; case 0x111d7603: /* 6 Port with Analog Mixer */ @@ -5031,6 +5032,9 @@ again: codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; } + if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP) + snd_hda_sequence_write_cache(codec, unmute_init); + spec->aloopback_mask = 0x50; spec->aloopback_shift = 0; -- 1.5.6.3

2 1

[alsa-devel] [PATCH] hda: 92hd71xxx disable unmute support for codecs that don't have input amps.
by Matthew Ranostay 26 Jan '09

26 Jan '09

Some revisions of the 92hd71xxx codec families don't have input amps on ports 0xa, 0xd and 0xf, so probe the widget caps on port 0xa and check for support, if found run snd_hda_sequence_write() on the stac92hd71xxx_unmute_core_init verb list. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- sound/pci/hda/patch_sigmatel.c | 20 ++++++++++++-------- 1 files changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 80a4c28..ffef091 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -858,26 +858,25 @@ static struct hda_verb stac92hd83xxx_core_init[] = { static struct hda_verb stac92hd71bxx_core_init[] = { /* set master volume and direct control */ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ - { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {} }; -#define HD_DISABLE_PORTF 2 +#define HD_DISABLE_PORTF 1 static struct hda_verb stac92hd71bxx_analog_core_init[] = { /* start of config #1 */ /* connect port 0f to audio mixer */ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, - /* unmute right and left channels for node 0x0f */ - { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* start of config #2 */ /* set master volume and direct control */ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* unmute right and left channels for nodes 0x0a, 0xd */ + {} +}; + +static struct hda_verb stac92hd71bxx_unmute_core_init[] = { + /* unmute right and left channels for nodes 0x0f, 0xa, 0x0d */ + { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {} @@ -4942,6 +4941,7 @@ static struct hda_input_mux stac92hd71bxx_dmux = { static int patch_stac92hd71bxx(struct hda_codec *codec) { struct sigmatel_spec *spec; + struct hda_verb *unmute_init = stac92hd71bxx_unmute_core_init; int err = 0; spec = kzalloc(sizeof(*spec), GFP_KERNEL); @@ -5015,6 +5015,7 @@ again: /* disable VSW */ spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF]; + unmute_init++; stac_change_pin_config(codec, 0xf, 0x40f000f0); break; case 0x111d7603: /* 6 Port with Analog Mixer */ @@ -5031,6 +5032,9 @@ again: codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; } + if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP) + snd_hda_sequence_write(codec, unmute_init); + spec->aloopback_mask = 0x50; spec->aloopback_shift = 0; -- 1.5.6.3

2 2

[alsa-devel] snd_pcm_set_params() bug?
by Christian Morales Vega 26 Jan '09

26 Jan '09

This code: #include <alsa/asoundlib.h> int main(){ snd_pcm_t *handle; if(snd_pcm_open(&handle, "default:1", SND_PCM_STREAM_PLAYBACK, 0) < 0) { exit(1); } if(snd_pcm_set_params(handle, SND_PCM_FORMAT_S16_LE, SND_PCM_ACCESS_RW_INTERLEAVED, 2, 24000, 1, 100000) < 0) { exit(2); } } fails with error "ALSA lib pcm.c:7193:(snd_pcm_set_params) Unable to get period size for PLAYBACK: Invalid argument" Disable resampling and you get "ALSA lib pcm.c:7142:(snd_pcm_set_params) Rate doesn't match (requested 24000Hz, get 0Hz)" (normal, I suppose). With other samplerates works fine. I'm using alsa-driver-kmp-default-1.0.16.20080602 and alsa-1.0.16.hg20080510 RPM packages from multimedia:audio openSUSE Build Service repository (from Takashi Iwai) and an AD1988 (hda-intel) codec. A XUbuntu user also has the same problem with an hda-intel codec (that I suppose uses stable 1.0.16). With a SB Live! in the same computer there are no problems. This was discovered with bsnes ( http://byuu.cinnamonpirate.com/bsnes/ - http://board.zsnes.com/phpBB2/viewtopic.php?t=11584 ) that can use openal, libao or alsa directly. The problem just happens when using 24KHz with alsa. So I suppose the problem is in the snd_pcm_set_params() itself, since openal and libao have not the problem but they set parameters "manually", not with snd_pcm_set_params().

3 2

[alsa-devel] [PATCH] Replace hardcoded hexa formating
by LCID Fire 26 Jan '09

26 Jan '09

The attached patch replaces the hardcoded hexa formats in usbaudio.c commit 079e4efba0603e7a32bf1d352d98c2b265b91cd6 Author: LCID Fire <lcid-fire(a)gmx.net> Date: Fri Jan 23 23:39:06 2009 +0100 Rather use printf format instead of hardcoding it. A next step would be to predefine certain formats. Signed-off-by: Andreas Bergmeier <lcid-fire(a)gmx.net> diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index bbd70d5..77ac5c9 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -1292,14 +1292,14 @@ static int init_usb_sample_rate(struct usb_device *dev, int iface, if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), SET_CUR, USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT, SAMPLING_FREQ_CONTROL << 8, ep, data, 3, 1000)) < 0) { - snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep 0x%x\n", + snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep %#x\n", dev->devnum, iface, fmt->altsetting, rate, ep); return err; } if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), GET_CUR, USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_IN, SAMPLING_FREQ_CONTROL << 8, ep, data, 3, 1000)) < 0) { - snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep 0x%x\n", + snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep %#x\n", dev->devnum, iface, fmt->altsetting, ep); return 0; /* some devices don't support reading */ } @@ -1468,7 +1468,7 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, channels = params_channels(hw_params); fmt = find_format(subs, format, rate, channels); if (!fmt) { - snd_printd(KERN_DEBUG "cannot set format: format = 0x%x, rate = %d, channels = %d\n", + snd_printd(KERN_DEBUG "cannot set format: format = %#x, rate = %d, channels = %d\n", format, rate, channels); return -EINVAL; } @@ -2160,7 +2160,7 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s fp = list_entry(p, struct audioformat, list); snd_iprintf(buffer, " Interface %d\n", fp->iface); snd_iprintf(buffer, " Altset %d\n", fp->altsetting); - snd_iprintf(buffer, " Format: 0x%x\n", fp->format); + snd_iprintf(buffer, " Format: %#x\n", fp->format); snd_iprintf(buffer, " Channels: %d\n", fp->channels); snd_iprintf(buffer, " Endpoint: %d %s (%s)\n", fp->endpoint & USB_ENDPOINT_NUMBER_MASK, @@ -2180,7 +2180,7 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s snd_iprintf(buffer, "\n"); } // snd_iprintf(buffer, " Max Packet Size = %d\n", fp->maxpacksize); - // snd_iprintf(buffer, " EP Attribute = 0x%x\n", fp->attributes); + // snd_iprintf(buffer, " EP Attribute = %#x\n", fp->attributes); } } @@ -2619,7 +2619,7 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, struct audioformat fp->format = SNDRV_PCM_FORMAT_MPEG; break; default: - snd_printd(KERN_INFO "%d:%u:%d : unknown format tag 0x%x is detected. processed as MPEG.\n", + snd_printd(KERN_INFO "%d:%u:%d : unknown format tag %#x is detected. processed as MPEG.\n", chip->dev->devnum, fp->iface, fp->altsetting, format); fp->format = SNDRV_PCM_FORMAT_MPEG; break; @@ -2817,7 +2817,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) continue; } - snd_printdd(KERN_INFO "%d:%u:%d: add audio endpoint 0x%x\n", dev->devnum, iface_no, altno, fp->endpoint); + snd_printdd(KERN_INFO "%d:%u:%d: add audio endpoint %#x\n", dev->devnum, iface_no, altno, fp->endpoint); err = add_audio_endpoint(chip, stream, fp); if (err < 0) { kfree(fp->rate_table);

2 1

[alsa-devel] opti92x datasheets
by Krzysztof Helt 26 Jan '09

26 Jan '09

Hi, I have found that Regensburg University mirrored Opti ftp site with drivers. The directories contain datasheets as well. Takashi can ask them if these datasheets can be copied into the alsa datasheet archive. The links are here: http://www-soft.uni-regensburg.de/soft/hardware/SOUND/OPTI/82C930/ http://www-soft.uni-regensburg.de/soft/hardware/SOUND/OPTI/82C929/ http://www-soft.uni-regensburg.de/soft/hardware/SOUND/OPTI/82C925/ http://www-soft.uni-regensburg.de/soft/hardware/SOUND/OPTI/82C924/ There are also some additional application notes. Regards, Krzysztof ---------------------------------------------------------------------- Kochanka - kobieta drugiej kategorii? sprawdz >>>http://link.interia.pl/f2026

2 1

[alsa-devel] Question on usbaudio.c: snd_usb_find_csint_desc
by LCID Fire 26 Jan '09

26 Jan '09

I was looking at 'snd_usb_find_csint_desc' and I really wonder why there is a while loop since it seems to me that the variables don't really change. Am I missing something?

2 1

[alsa-devel] Enumerating PCM devices
by Bankim Bhavsar 25 Jan '09

25 Jan '09

Hi Alsa developers/users, I'm using snd_device_name_hint("pcm"..) and then snd_device_name_get_hint(.."NAME") to get the NAME parameter of pcm devices.

1 0

Re: [alsa-devel] [PATCH] Turtle Beach Multisound Classic/Pinnacle driver (4th rev)
by Krzysztof Helt 24 Jan '09

24 Jan '09

On Sat, 24 Jan 2009 13:35:28 +0100 Krzysztof Helt <krzysztof.h1(a)poczta.fm> wrote: > From: Krzysztof Helt <krzysztof.h1(a)wp.pl> > > + .buffer_bytes_max = 0x3000, > + .period_bytes_min = 0x40, > + .period_bytes_max = 0x1800, > + .periods_min = 2, > + .periods_max = 3, I have investigated the max_periods number and at least some code called from the interrupt assumes no more then 3 periods in the buffer. I do not understand the code enough to change this. Regards, Krzysztof ---------------------------------------------------------------------- Zobacz gry internetowe dla dzieci! sprawdz >>>http://link.interia.pl/f202c

1 0

[PATCH] Add device and control number so it's apparent for which control this error is showing up.
by Andreas Bergmeier 24 Jan '09

24 Jan '09

Signed-off-by: Andreas Bergmeier <lcid-fire(a)gmx.net> --- sound/usb/usbaudio.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 037435d..e578a5c 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2867,12 +2867,12 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) /* find audiocontrol interface */ host_iface = &usb_ifnum_to_if(dev, ctrlif)->altsetting[0]; if (!(p1 = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, NULL, HEADER))) { - snd_printk(KERN_ERR "cannot find HEADER\n"); + snd_printk(KERN_ERR "%d:%u: cannot find HEADER\n", dev->devnum, ctrlif); return -EINVAL; } if (! p1[7] || p1[0] < 8 + p1[7]) { - snd_printk(KERN_ERR "invalid HEADER\n"); + snd_printk(KERN_ERR "%d:%u: invalid HEADER\n", dev->devnum, ctrlif); return -EINVAL; } -- 1.5.6.3 --------------050806070602090407040706 Content-Type: text/x-patch; name="0001-Print-device-number-which-did-not-support-low-speed.patch" Content-Transfer-Encoding: 7bit Content-Disposition: inline; filename*0="0001-Print-device-number-which-did-not-support-low-speed.pat"; filename*1="ch"

1 0

[PATCH] Print device number which did not support low speed audio streaming
by Andreas Bergmeier 24 Jan '09

24 Jan '09

Signed-off-by: Andreas Bergmeier <lcid-fire(a)gmx.net> --- sound/usb/usbaudio.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 8b8eb9e..2cfb8a3 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2929,7 +2929,7 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) continue; } if (snd_usb_get_speed(dev) == USB_SPEED_LOW) { - snd_printk(KERN_ERR "low speed audio streaming not supported\n"); + snd_printk(KERN_ERR "%d: low speed audio streaming not supported\n", dev->devnum); continue; } if (! parse_audio_endpoints(chip, j)) { -- 1.5.6.3 --------------050806070602090407040706 Content-Type: text/x-patch; name="0001-Replaced-return-NULL-lines-by-break-in-snd_usb_fin.patch" Content-Transfer-Encoding: 7bit Content-Disposition: inline; filename*0="0001-Replaced-return-NULL-lines-by-break-in-snd_usb_fin.patc"; filename*1="h"

1 0

[PATCH] Replaced 'return NULL' lines by break in snd_usb_find_desc so there is only one default error.
by Andreas Bergmeier 24 Jan '09

24 Jan '09

Signed-off-by: Andreas Bergmeier <lcid-fire(a)gmx.net> --- sound/usb/usbaudio.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index d35b51c..8b8eb9e 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2054,10 +2054,10 @@ void *snd_usb_find_desc(void *descstart, int desclen, void *after, u8 dtype) end = p + desclen; for (; p < end;) { if (p[0] < 2) - return NULL; + break; next = p + p[0]; if (next > end) - return NULL; + break; if (p[1] == dtype && (!after || (void *)p > after)) { return p; } -- 1.5.6.3 --------------050806070602090407040706--

1 0

[alsa-devel] snd_pcm_avail_update() needed before snd_pcm_delay() with hda-intel
by Christian Morales Vega 24 Jan '09

24 Jan '09

snd_pcm_avail_update() documentation says: "Using of this function is useless for the standard read/write operations. Use it only for mmap access. See to snd_pcm_delay." snd_pcm_delay() documentation says: "Note this function does not update the actual r/w pointer for applications. The function snd_pcm_avail_update() have to be called before any begin+commit operation." My english is far from perfect, and so the problem can be that I missunderstood the docs but: 1- "Using of this function"? Should not be "Use of this function"? 2- Is snd_pcm_avail_update() really useless when not using mmap? I have a SB Live! where snd_pcm_delay() always returns the correct value, without any need for snd_pcm_avail_update(). But the same code in the same machine with an hda-intel AD1988 needs a call to snd_pcm_avail_update() before snd_pcm_delay() to give the correct result. Without it snd_pcm_delay() always returns the value that *was* correct in the last snd_pcm_writei() call. So should the "application frame position" be always updated, or only in read/write operations? In the first case AD1988 fails, in the second case perhaps SB Live! is making too mucho work ;-)

4 9

[alsa-devel] [PATCH v2 2.6.30] ASoC: improve I2C initialization code in CS4270 driver
by Timur Tabi 24 Jan '09

24 Jan '09

Further improvements in the I2C initialization sequence of the CS4270 driver. All ASoC initialization is now done in the I2C probe function. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/codecs/cs4270.c | 252 ++++++++++++++++++++------------------------- 1 files changed, 113 insertions(+), 139 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 2e4ce04..71986e1 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -29,12 +29,6 @@ #include <sound/initval.h> #include <linux/i2c.h> -/* Private data for the CS4270 */ -struct cs4270_private { - unsigned int mclk; /* Input frequency of the MCLK pin */ - unsigned int mode; /* The mode (I2S or left-justified) */ -}; - /* * The codec isn't really big-endian or little-endian, since the I2S * interface requires data to be sent serially with the MSbit first. @@ -107,6 +101,14 @@ struct cs4270_private { #define CS4270_MUTE_DAC_A 0x01 #define CS4270_MUTE_DAC_B 0x02 +/* Private data for the CS4270 */ +struct cs4270_private { + struct snd_soc_codec codec; + u8 reg_cache[CS4270_NUMREGS]; + unsigned int mclk; /* Input frequency of the MCLK pin */ + unsigned int mode; /* The mode (I2S or left-justified) */ +}; + /* * Clock Ratio Selection for Master Mode with I2C enabled * @@ -502,6 +504,31 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = { */ static struct snd_soc_device *cs4270_socdev; +struct snd_soc_dai cs4270_dai = { + .name = "cs4270", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = 0, + .formats = CS4270_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = 0, + .formats = CS4270_FORMATS, + }, + .ops = { + .hw_params = cs4270_hw_params, + .set_sysclk = cs4270_set_dai_sysclk, + .set_fmt = cs4270_set_dai_fmt, + .digital_mute = cs4270_mute, + }, +}; +EXPORT_SYMBOL_GPL(cs4270_dai); + /* * Initialize the I2C interface of the CS4270 * @@ -515,47 +542,52 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, const struct i2c_device_id *id) { struct snd_soc_device *socdev = cs4270_socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec; + struct cs4270_private *cs4270; int i; int ret = 0; - /* Probing all possible addresses has one drawback: if there are - multiple CS4270s on the bus, then you cannot specify which - socdev is matched with which CS4270. For now, we just reject - this I2C device if the socdev already has one attached. */ - if (codec->control_data) - return -ENODEV; - - /* Note: codec_dai->codec is NULL here */ - - codec->reg_cache = kzalloc(CS4270_NUMREGS, GFP_KERNEL); - if (!codec->reg_cache) { - printk(KERN_ERR "cs4270: could not allocate register cache\n"); - ret = -ENOMEM; - goto error; - } - /* Verify that we have a CS4270 */ ret = i2c_smbus_read_byte_data(i2c_client, CS4270_CHIPID); if (ret < 0) { printk(KERN_ERR "cs4270: failed to read I2C\n"); - goto error; + return ret; } /* The top four bits of the chip ID should be 1100. */ if ((ret & 0xF0) != 0xC0) { - /* The device at this address is not a CS4270 codec */ - ret = -ENODEV; - goto error; + printk(KERN_ERR "cs4270: device at addr %X is not a CS4270\n", + i2c_client->addr); + return -ENODEV; } printk(KERN_INFO "cs4270: found device at I2C address %X\n", i2c_client->addr); printk(KERN_INFO "cs4270: hardware revision %X\n", ret & 0xF); + /* Allocate enough space for the snd_soc_codec structure + and our private data together. */ + cs4270 = kzalloc(sizeof(struct cs4270_private), GFP_KERNEL); + if (!cs4270) { + printk(KERN_ERR "cs4270: Could not allocate codec structure\n"); + return -ENOMEM; + } + codec = &cs4270->codec; + socdev->codec = codec; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->name = "CS4270"; + codec->owner = THIS_MODULE; + codec->dai = &cs4270_dai; + codec->num_dai = 1; + codec->private_data = cs4270; codec->control_data = i2c_client; codec->read = cs4270_read_reg_cache; codec->write = cs4270_i2c_write; + codec->reg_cache = cs4270->reg_cache; codec->reg_cache_size = CS4270_NUMREGS; /* The I2C interface is set up, so pre-fill our register cache */ @@ -563,35 +595,72 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, ret = cs4270_fill_cache(codec); if (ret < 0) { printk(KERN_ERR "cs4270: failed to fill register cache\n"); - goto error; + goto error_free_codec; + } + + /* Register PCMs */ + + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "cs4270: failed to create PCMs\n"); + goto error_free_codec; } /* Add the non-DAPM controls */ for (i = 0; i < ARRAY_SIZE(cs4270_snd_controls); i++) { - struct snd_kcontrol *kctrl = - snd_soc_cnew(&cs4270_snd_controls[i], codec, NULL); + struct snd_kcontrol *kctrl; + + kctrl = snd_soc_cnew(&cs4270_snd_controls[i], codec, NULL); + if (!kctrl) { + printk(KERN_ERR "cs4270: error creating control '%s'\n", + cs4270_snd_controls[i].name); + ret = -ENOMEM; + goto error_free_pcms; + } ret = snd_ctl_add(codec->card, kctrl); - if (ret < 0) - goto error; + if (ret < 0) { + printk(KERN_ERR "cs4270: error adding control '%s'\n", + cs4270_snd_controls[i].name); + goto error_free_pcms; + } } - i2c_set_clientdata(i2c_client, codec); + /* Initialize the SOC device */ + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + printk(KERN_ERR "cs4270: failed to register card\n"); + goto error_free_pcms;; + } + + i2c_set_clientdata(i2c_client, socdev); return 0; -error: - codec->control_data = NULL; +error_free_pcms: + snd_soc_free_pcms(socdev); - kfree(codec->reg_cache); - codec->reg_cache = NULL; - codec->reg_cache_size = 0; +error_free_codec: + kfree(cs4270); return ret; } -static const struct i2c_device_id cs4270_id[] = { +static int cs4270_i2c_remove(struct i2c_client *i2c_client) +{ + struct snd_soc_device *socdev = i2c_get_clientdata(i2c_client); + struct snd_soc_codec *codec = socdev->codec; + struct cs4270_private *cs4270 = codec->private_data; + + snd_soc_free_pcms(socdev); + kfree(cs4270); + + return 0; +} + +static struct i2c_device_id cs4270_id[] = { {"cs4270", 0}, {} }; @@ -604,27 +673,9 @@ static struct i2c_driver cs4270_i2c_driver = { }, .id_table = cs4270_id, .probe = cs4270_i2c_probe, + .remove = cs4270_i2c_remove, }; -struct snd_soc_dai cs4270_dai = { - .name = "CS4270", - .playback = { - .stream_name = "Playback", - .channels_min = 1, - .channels_max = 2, - .rates = 0, - .formats = CS4270_FORMATS, - }, - .capture = { - .stream_name = "Capture", - .channels_min = 1, - .channels_max = 2, - .rates = 0, - .formats = CS4270_FORMATS, - }, -}; -EXPORT_SYMBOL_GPL(cs4270_dai); - /* * ASoC probe function * @@ -633,94 +684,15 @@ EXPORT_SYMBOL_GPL(cs4270_dai); */ static int cs4270_probe(struct platform_device *pdev) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec; - int ret = 0; - - printk(KERN_INFO "CS4270 ALSA SoC Codec\n"); - - /* Allocate enough space for the snd_soc_codec structure - and our private data together. */ - codec = kzalloc(ALIGN(sizeof(struct snd_soc_codec), 4) + - sizeof(struct cs4270_private), GFP_KERNEL); - if (!codec) { - printk(KERN_ERR "cs4270: Could not allocate codec structure\n"); - return -ENOMEM; - } - - mutex_init(&codec->mutex); - INIT_LIST_HEAD(&codec->dapm_widgets); - INIT_LIST_HEAD(&codec->dapm_paths); - - codec->name = "CS4270"; - codec->owner = THIS_MODULE; - codec->dai = &cs4270_dai; - codec->num_dai = 1; - codec->private_data = (void *) codec + - ALIGN(sizeof(struct snd_soc_codec), 4); - - socdev->codec = codec; - - /* Register PCMs */ - - ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); - if (ret < 0) { - printk(KERN_ERR "cs4270: failed to create PCMs\n"); - goto error_free_codec; - } - - cs4270_socdev = socdev; - - ret = i2c_add_driver(&cs4270_i2c_driver); - if (ret) { - printk(KERN_ERR "cs4270: failed to attach driver"); - goto error_free_pcms; - } - - /* Did we find a CS4270 on the I2C bus? */ - if (!codec->control_data) { - printk(KERN_ERR "cs4270: failed to attach driver"); - goto error_del_driver; - } + cs4270_socdev = platform_get_drvdata(pdev);; - /* Initialize codec ops */ - cs4270_dai.ops.hw_params = cs4270_hw_params; - cs4270_dai.ops.set_sysclk = cs4270_set_dai_sysclk; - cs4270_dai.ops.set_fmt = cs4270_set_dai_fmt; - cs4270_dai.ops.digital_mute = cs4270_mute; - - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "cs4270: failed to register card\n"); - goto error_del_driver; - } - - return 0; - -error_del_driver: - i2c_del_driver(&cs4270_i2c_driver); - -error_free_pcms: - snd_soc_free_pcms(socdev); - -error_free_codec: - kfree(socdev->codec); - socdev->codec = NULL; - - return ret; + return i2c_add_driver(&cs4270_i2c_driver); } static int cs4270_remove(struct platform_device *pdev) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - - snd_soc_free_pcms(socdev); - i2c_del_driver(&cs4270_i2c_driver); - kfree(socdev->codec); - socdev->codec = NULL; - return 0; } @@ -738,6 +710,8 @@ EXPORT_SYMBOL_GPL(soc_codec_device_cs4270); static int __init cs4270_init(void) { + printk(KERN_INFO "Cirrus Logic CS4270 ALSA SoC Codec Driver\n"); + return snd_soc_register_dai(&cs4270_dai); } module_init(cs4270_init); -- 1.5.5

2 1

[alsa-devel] ASoC updats for 2.6.30
by Mark Brown 23 Jan '09

23 Jan '09

The following changes since commit 56692396447ca3fee061c26f22590a1046fd2d6b: Mark Brown (1): Merge branch 'for-2.6.29' into for-2.6.30 are available in the git repository at: ssh://master.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 (the usual git URI should work eventually as well but the mirroring appears to be stuck right now and it's past 5pm on a Friday.) Mark Brown (7): ASoC: Fix merge with PXA tree ASoC: Remove unneeded e7x0 inclusion of pxa-regs.h and hardware.h ASoC: Configure SSP port PLL for Zylonite ASoC: Fix spurious codec driver dependencies Merge branch 'for-2.6.29' into for-2.6.30 ASoC: Include header file in cs4270 and wm9705 ASoC: Fix L3 bus handling in Kconfig Peter Ujfalusi (1): ASoC: Add missing comma to SND_SOC_DAPM_SWITCH_E in soc-dapm.h Steve Sakoman (1): ASoC: Complete Beagleboard support Timur Tabi (1): ASoC: remove stand-alone mode support from CS4270 codec driver include/sound/soc-dapm.h | 2 +- sound/soc/codecs/Kconfig | 5 +-- sound/soc/codecs/cs4270.c | 90 +++++++++++++------------------------------ sound/soc/codecs/wm9705.c | 2 + sound/soc/fsl/Kconfig | 3 +- sound/soc/omap/Kconfig | 14 ++++++- sound/soc/omap/Makefile | 2 + sound/soc/pxa/e740_wm9705.c | 2 - sound/soc/pxa/e750_wm9705.c | 2 - sound/soc/pxa/e800_wm9712.c | 5 +-- sound/soc/pxa/zylonite.c | 8 ++++ sound/soc/s3c24xx/Kconfig | 1 + 12 files changed, 57 insertions(+), 79 deletions(-)

2 10

[alsa-devel] ASoC update for 2.6.29
by Mark Brown 23 Jan '09

23 Jan '09

The following changes since commit 8a9dee59a345f96757dd45699de1c4182d8bf9a9: Timur Tabi (1): ASoC: fix registration of the SoC card in the Freescale MPC8610 drivers are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.29 Peter Ujfalusi (1): ASoC: Add missing comma to SND_SOC_DAPM_SWITCH_E in soc-dapm.h include/sound/soc-dapm.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-)

2 1

[alsa-devel] Acer Aspire 8920g / snd-hda-intel / ALC889 / surround sound
by Sergey 'Jin' Bostandzhyan 23 Jan '09

23 Jan '09

Hi, there's been a thread about a similar issue around 2008-04, it ended with a working front left/right setup, which is of course better than no sound, but it would be really nice to be able to use all available speakers. The notebook has 5.1 surround, but all speakers except front left and front right stay silent. I'm running Fedora 10, 2.6.27.9-159.fc10.x86_64, # cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.18a. # cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xfc300000 irq 22 1 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xc7eec000 irq 17 When I select model "auto", then I get a working front left and front right speaker. When I select model "acer-aspire", everything is mute and I have to use the hda-verb utility to get some sound out of the front speakers, so right now I am using the "auto" option. I see a segfault in alsactl when I look at dmesg, not sure if this is related to the problem: HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 HDA Intel 0000:00:1b.0: setting latency timer to 64 ALSA sound/pci/hda/hda_codec.c:2587: hda_codec: model 'auto' is selected ALSA sound/pci/hda/hda_codec.c:3303: autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:3307: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:3311: hp_outs=1 (0x15/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:3312: mono: mono_out=0x0 ALSA sound/pci/hda/hda_codec.c:3320: inputs: mic=0x12, fmic=0x18, line=0x1a, fline=0x0, cd=0x0, aux=0x0 HDA Intel 0000:01:00.1: PCI INT B -> GSI 17 (level, low) -> IRQ 17 HDA Intel 0000:01:00.1: setting latency timer to 64 alsactl[1631]: segfault at 0 ip 00000000004106e7 sp 00007fffb7682fa0 error 4 in alsactl[400000+14000] alsactl[1624]: segfault at 0 ip 00000000004106e7 sp 00007fff4ba9e3c0 error 4 in alsactl[400000+14000] The full output of the alsa-info script is here: http://www.alsa-project.org/db/?f=405364959a8f52c85f045f35fa1f3bf4536b13bc I have a working setup to compile the alsa-driver, so I'm ready to test whatever is needed :) Any ideas? Kind regards, Jin

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[alsa-devel] [PATCH] Turtle Beach Multisound Classic/Pinnacle driver (3rd rev)
by Krzysztof Helt 23 Jan '09

23 Jan '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This is driver for Turtle Beach Multisound cards: Classic, Fiji and Pinnacle. Tested pcm playback and recording and MIDI playback on Multisound Pinnacle. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- All issues pointed by Takashi are resolved. I moved more code to the common part and made a common library. diff -urpN linux-rc5/sound/isa/msnd/Makefile linux-git/sound/isa/msnd/Makefile --- linux-rc5/sound/isa/msnd/Makefile 1970-01-01 01:00:00.000000000 +0100 +++ linux-git/sound/isa/msnd/Makefile 2009-01-22 20:09:26.126460759 +0100 @@ -0,0 +1,9 @@ + +snd-msnd-lib-objs := msnd.o msnd_midi.o +snd-msnd-pinnacle-objs := msnd_pinnacle.o msnd_pinnacle_mixer.o +snd-msnd-classic-objs := msnd_classic.o msnd_pinnacle_mixer.o + +# Toplevel Module Dependency +obj-$(CONFIG_SND_MSND_PINNACLE) += snd-msnd-pinnacle.o snd-msnd-lib.o +obj-$(CONFIG_SND_MSND_CLASSIC) += snd-msnd-classic.o snd-msnd-lib.o + diff -urpN linux-rc5/sound/isa/msnd/msnd.c linux-git/sound/isa/msnd/msnd.c --- linux-rc5/sound/isa/msnd/msnd.c 1970-01-01 01:00:00.000000000 +0100 +++ linux-git/sound/isa/msnd/msnd.c 2009-01-22 21:19:43.283202276 +0100 @@ -0,0 +1,699 @@ +/********************************************************************* + * + * 2002/06/30 Karsten Wiese: + * removed kernel-version dependencies. + * ripped from linux kernel 2.4.18 (OSS Implementation) by me. + * In the OSS Version, this file is compiled to a separate MODULE, + * that is used by the pinnacle and the classic driver. + * since there is no classic driver for alsa yet (i dont have a classic + * & writing one blindfold is difficult) this file's object is statically + * linked into the pinnacle-driver-module for now. look for the string + * "uncomment this to make this a module again" + * to do guess what. + * + * the following is a copy of the 2.4.18 OSS FREE file-heading comment: + * + * msnd.c - Driver Base + * + * Turtle Beach MultiSound Sound Card Driver for Linux + * + * Copyright (C) 1998 Andrew Veliath + * + * 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., 675 Mass Ave, Cambridge, MA 02139, USA. + * + ********************************************************************/ + +#include <linux/kernel.h> +#include <linux/types.h> +#include <linux/interrupt.h> +#include <linux/io.h> +#include <linux/fs.h> +#include <linux/delay.h> + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> + +#include "msnd.h" + +#define LOGNAME "msnd" + + +void snd_msnd_init_queue(void *base, int start, int size) +{ + writew(PCTODSP_BASED(start), base + JQS_wStart); + writew(PCTODSP_OFFSET(size) - 1, base + JQS_wSize); + writew(0, base + JQS_wHead); + writew(0, base + JQS_wTail); +} +EXPORT_SYMBOL(snd_msnd_init_queue); + +static int snd_msnd_wait_TXDE(struct snd_msnd *dev) +{ + unsigned int io = dev->io; + int timeout = 1000; + + while (timeout-- > 0) + if (inb(io + HP_ISR) & HPISR_TXDE) + return 0; + + return -EIO; +} + +static int snd_msnd_wait_HC0(struct snd_msnd *dev) +{ + unsigned int io = dev->io; + int timeout = 1000; + + while (timeout-- > 0) + if (!(inb(io + HP_CVR) & HPCVR_HC)) + return 0; + + return -EIO; +} + +int snd_msnd_send_dsp_cmd(struct snd_msnd *dev, u8 cmd) +{ + unsigned long flags; + + spin_lock_irqsave(&dev->lock, flags); + if (snd_msnd_wait_HC0(dev) == 0) { + outb(cmd, dev->io + HP_CVR); + spin_unlock_irqrestore(&dev->lock, flags); + return 0; + } + spin_unlock_irqrestore(&dev->lock, flags); + + snd_printd(KERN_ERR LOGNAME ": Send DSP command timeout\n"); + + return -EIO; +} +EXPORT_SYMBOL(snd_msnd_send_dsp_cmd); + +int snd_msnd_send_word(struct snd_msnd *dev, unsigned char high, + unsigned char mid, unsigned char low) +{ + unsigned int io = dev->io; + + if (snd_msnd_wait_TXDE(dev) == 0) { + outb(high, io + HP_TXH); + outb(mid, io + HP_TXM); + outb(low, io + HP_TXL); + return 0; + } + + snd_printd(KERN_ERR LOGNAME ": Send host word timeout\n"); + + return -EIO; +} +EXPORT_SYMBOL(snd_msnd_send_word); + +int snd_msnd_upload_host(struct snd_msnd *dev, const u8 *bin, int len) +{ + int i; + + if (len % 3 != 0) { + snd_printk(LOGNAME ": Upload host data not multiple of 3!\n"); + return -EINVAL; + } + + for (i = 0; i < len; i += 3) + if (snd_msnd_send_word(dev, bin[i], bin[i + 1], bin[i + 2])) + return -EIO; + + inb(dev->io + HP_RXL); + inb(dev->io + HP_CVR); + + return 0; +} +EXPORT_SYMBOL(snd_msnd_upload_host); + +int snd_msnd_enable_irq(struct snd_msnd *dev) +{ + unsigned long flags; + + if (dev->irq_ref++) + return 0; + + snd_printdd(LOGNAME ": Enabling IRQ\n"); + + spin_lock_irqsave(&dev->lock, flags); + if (snd_msnd_wait_TXDE(dev) == 0) { + outb(inb(dev->io + HP_ICR) | HPICR_TREQ, dev->io + HP_ICR); + if (dev->type == msndClassic) + outb(dev->irqid, dev->io + HP_IRQM); + + outb(inb(dev->io + HP_ICR) & ~HPICR_TREQ, dev->io + HP_ICR); + outb(inb(dev->io + HP_ICR) | HPICR_RREQ, dev->io + HP_ICR); + enable_irq(dev->irq); + snd_msnd_init_queue(dev->DSPQ, dev->dspq_data_buff, + dev->dspq_buff_size); + spin_unlock_irqrestore(&dev->lock, flags); + return 0; + } + spin_unlock_irqrestore(&dev->lock, flags); + + snd_printd(KERN_ERR LOGNAME ": Enable IRQ failed\n"); + + return -EIO; +} +EXPORT_SYMBOL(snd_msnd_enable_irq); + +int snd_msnd_disable_irq(struct snd_msnd *dev) +{ + unsigned long flags; + + if (--dev->irq_ref > 0) + return 0; + + if (dev->irq_ref < 0) + snd_printd(KERN_WARNING LOGNAME ": IRQ ref count is %d\n", + dev->irq_ref); + + snd_printdd(LOGNAME ": Disabling IRQ\n"); + + spin_lock_irqsave(&dev->lock, flags); + if (snd_msnd_wait_TXDE(dev) == 0) { + outb(inb(dev->io + HP_ICR) & ~HPICR_RREQ, dev->io + HP_ICR); + if (dev->type == msndClassic) + outb(HPIRQ_NONE, dev->io + HP_IRQM); + disable_irq(dev->irq); + spin_unlock_irqrestore(&dev->lock, flags); + return 0; + } + spin_unlock_irqrestore(&dev->lock, flags); + + snd_printd(KERN_ERR LOGNAME ": Disable IRQ failed\n"); + + return -EIO; +} +EXPORT_SYMBOL(snd_msnd_disable_irq); + +static inline long get_play_delay_jiffies(struct snd_msnd *chip, long size) +{ + long tmp = (size * HZ * chip->play_sample_size) / 8; + return tmp / (chip->play_sample_rate * chip->play_channels); +} + +static void snd_msnd_dsp_write_flush(struct snd_msnd *chip) +{ + if (!(chip->mode & FMODE_WRITE) || !test_bit(F_WRITING, &chip->flags)) + return; + set_bit(F_WRITEFLUSH, &chip->flags); +/* interruptible_sleep_on_timeout( + &chip->writeflush, + get_play_delay_jiffies(&chip, chip->DAPF.len));*/ + clear_bit(F_WRITEFLUSH, &chip->flags); + if (!signal_pending(current)) + schedule_timeout_interruptible( + get_play_delay_jiffies(chip, chip->play_period_bytes)); + clear_bit(F_WRITING, &chip->flags); +} + +void snd_msnd_dsp_halt(struct snd_msnd *chip, struct file *file) +{ + if ((file ? file->f_mode : chip->mode) & FMODE_READ) { + clear_bit(F_READING, &chip->flags); + snd_msnd_send_dsp_cmd(chip, HDEX_RECORD_STOP); + snd_msnd_disable_irq(chip); + if (file) { + snd_printd(KERN_INFO LOGNAME + ": Stopping read for %p\n", file); + chip->mode &= ~FMODE_READ; + } + clear_bit(F_AUDIO_READ_INUSE, &chip->flags); + } + if ((file ? file->f_mode : chip->mode) & FMODE_WRITE) { + if (test_bit(F_WRITING, &chip->flags)) { + snd_msnd_dsp_write_flush(chip); + snd_msnd_send_dsp_cmd(chip, HDEX_PLAY_STOP); + } + snd_msnd_disable_irq(chip); + if (file) { + snd_printd(KERN_INFO + LOGNAME ": Stopping write for %p\n", file); + chip->mode &= ~FMODE_WRITE; + } + clear_bit(F_AUDIO_WRITE_INUSE, &chip->flags); + } +} +EXPORT_SYMBOL(snd_msnd_dsp_halt); + + +int snd_msnd_DARQ(struct snd_msnd *chip, int bank) +{ + int /*size, n,*/ timeout = 3; + u16 wTmp; + /* void *DAQD; */ + + /* Increment the tail and check for queue wrap */ + wTmp = readw(chip->DARQ + JQS_wTail) + PCTODSP_OFFSET(DAQDS__size); + if (wTmp > readw(chip->DARQ + JQS_wSize)) + wTmp = 0; + while (wTmp == readw(chip->DARQ + JQS_wHead) && timeout--) + udelay(1); + + if (chip->capturePeriods == 2) { + void *pDAQ = chip->mappedbase + DARQ_DATA_BUFF + + bank * DAQDS__size + DAQDS_wStart; + unsigned short offset = 0x3000 + chip->capturePeriodBytes; + + if (readw(pDAQ) != PCTODSP_BASED(0x3000)) + offset = 0x3000; + writew(PCTODSP_BASED(offset), pDAQ); + } + + writew(wTmp, chip->DARQ + JQS_wTail); + +#if 0 + /* Get our digital audio queue struct */ + DAQD = bank * DAQDS__size + chip->mappedbase + DARQ_DATA_BUFF; + + /* Get length of data */ + size = readw(DAQD + DAQDS_wSize); + + /* Read data from the head (unprotected bank 1 access okay + since this is only called inside an interrupt) */ + outb(HPBLKSEL_1, chip->io + HP_BLKS); + if ((n = msnd_fifo_write( + &chip->DARF, + (char *)(chip->base + bank * DAR_BUFF_SIZE), + size, 0)) <= 0) { + outb(HPBLKSEL_0, chip->io + HP_BLKS); + return n; + } + outb(HPBLKSEL_0, chip->io + HP_BLKS); +#endif + + return 1; +} +EXPORT_SYMBOL(snd_msnd_DARQ); + +int snd_msnd_DAPQ(struct snd_msnd *chip, int start) +{ + u16 DAPQ_tail; + int protect = start, nbanks = 0; + void *DAQD; + static int play_banks_submitted; + /* unsigned long flags; + spin_lock_irqsave(&chip->lock, flags); not necessary */ + + DAPQ_tail = readw(chip->DAPQ + JQS_wTail); + while (DAPQ_tail != readw(chip->DAPQ + JQS_wHead) || start) { + int bank_num = DAPQ_tail / PCTODSP_OFFSET(DAQDS__size); + + if (start) { + start = 0; + play_banks_submitted = 0; + } + + /* Get our digital audio queue struct */ + DAQD = bank_num * DAQDS__size + chip->mappedbase + + DAPQ_DATA_BUFF; + + /* Write size of this bank */ + writew(chip->play_period_bytes, DAQD + DAQDS_wSize); + if (play_banks_submitted < 3) + ++play_banks_submitted; + else if (chip->playPeriods == 2) { + unsigned short offset = chip->play_period_bytes; + + if (readw(DAQD + DAQDS_wStart) != PCTODSP_BASED(0x0)) + offset = 0; + + writew(PCTODSP_BASED(offset), DAQD + DAQDS_wStart); + } + ++nbanks; + + /* Then advance the tail */ + /* + if (protect) + snd_printd(KERN_INFO "B %X %lX\n", + bank_num, xtime.tv_usec); + */ + + DAPQ_tail = (++bank_num % 3) * PCTODSP_OFFSET(DAQDS__size); + writew(DAPQ_tail, chip->DAPQ + JQS_wTail); + /* Tell the DSP to play the bank */ + snd_msnd_send_dsp_cmd(chip, HDEX_PLAY_START); + if (protect) + if (2 == bank_num) + break; + } + /* + if (protect) + snd_printd(KERN_INFO "%lX\n", xtime.tv_usec); + */ + /* spin_unlock_irqrestore(&chip->lock, flags); not necessary */ + return nbanks; +} +EXPORT_SYMBOL(snd_msnd_DAPQ); + +static void snd_msnd_play_reset_queue(struct snd_msnd *chip, + unsigned int pcm_periods, + unsigned int pcm_count) +{ + int n; + void *pDAQ = chip->mappedbase + DAPQ_DATA_BUFF; + + chip->last_playbank = -1; + chip->playLimit = pcm_count * (pcm_periods - 1); + chip->playPeriods = pcm_periods; + writew(PCTODSP_OFFSET(0 * DAQDS__size), chip->DAPQ + JQS_wHead); + writew(PCTODSP_OFFSET(0 * DAQDS__size), chip->DAPQ + JQS_wTail); + + chip->play_period_bytes = pcm_count; + + for (n = 0; n < pcm_periods; ++n, pDAQ += DAQDS__size) { + writew(PCTODSP_BASED((u32)(pcm_count * (n % pcm_periods))), + pDAQ + DAQDS_wStart); + writew(0, pDAQ + DAQDS_wSize); + writew(1, pDAQ + DAQDS_wFormat); + writew(chip->play_sample_size, pDAQ + DAQDS_wSampleSize); + writew(chip->play_channels, pDAQ + DAQDS_wChannels); + writew(chip->play_sample_rate, pDAQ + DAQDS_wSampleRate); + writew(HIMT_PLAY_DONE * 0x100 + n, pDAQ + DAQDS_wIntMsg); + writew(n, pDAQ + DAQDS_wFlags); + } +} + +static void snd_msnd_capture_reset_queue(struct snd_msnd *chip, + unsigned int pcm_periods, + unsigned int pcm_count) +{ + int n; + void *pDAQ; + /* unsigned long flags; */ + + /* snd_msnd_init_queue(chip->DARQ, DARQ_DATA_BUFF, DARQ_BUFF_SIZE); */ + + chip->last_recbank = 2; + chip->captureLimit = pcm_count * (pcm_periods - 1); + chip->capturePeriods = pcm_periods; + writew(PCTODSP_OFFSET(0 * DAQDS__size), chip->DARQ + JQS_wHead); + writew(PCTODSP_OFFSET(chip->last_recbank * DAQDS__size), + chip->DARQ + JQS_wTail); + + /* Critical section: bank 1 access. this is how the OSS driver does it: + spin_lock_irqsave(&chip->lock, flags); + outb(HPBLKSEL_1, chip->io + HP_BLKS); + memset_io(chip->mappedbase, 0, DAR_BUFF_SIZE * 3); + outb(HPBLKSEL_0, chip->io + HP_BLKS); + spin_unlock_irqrestore(&chip->lock, flags);*/ + + chip->capturePeriodBytes = pcm_count; + snd_printdd("snd_msnd_capture_reset_queue() %i\n", pcm_count); + + pDAQ = chip->mappedbase + DARQ_DATA_BUFF; + + for (n = 0; n < pcm_periods; ++n, pDAQ += DAQDS__size) { + u32 tmp = pcm_count * (n % pcm_periods); + + writew(PCTODSP_BASED(tmp + 0x3000), pDAQ + DAQDS_wStart); + writew(pcm_count, pDAQ + DAQDS_wSize); + writew(1, pDAQ + DAQDS_wFormat); + writew(chip->capture_sample_size, pDAQ + DAQDS_wSampleSize); + writew(chip->capture_channels, pDAQ + DAQDS_wChannels); + writew(chip->capture_sample_rate, pDAQ + DAQDS_wSampleRate); + writew(HIMT_RECORD_DONE * 0x100 + n, pDAQ + DAQDS_wIntMsg); + writew(n, pDAQ + DAQDS_wFlags); + } +} + +static struct snd_pcm_hardware snd_msnd_playback = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID, + .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = 0x3000, + .period_bytes_min = 0x40, + .period_bytes_max = 0x1800, + .periods_min = 2, + .periods_max = 3, + .fifo_size = 0, +}; + +static struct snd_pcm_hardware snd_msnd_capture = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID, + .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = 0x3000, + .period_bytes_min = 0x40, + .period_bytes_max = 0x1800, + .periods_min = 2, + .periods_max = 3, + .fifo_size = 0, +}; + + +static int snd_msnd_playback_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + set_bit(F_AUDIO_WRITE_INUSE, &chip->flags); + clear_bit(F_WRITING, &chip->flags); + snd_msnd_enable_irq(chip); + + runtime->dma_area = chip->mappedbase; + runtime->dma_bytes = 0x3000; + + chip->playback_substream = substream; + runtime->hw = snd_msnd_playback; + return 0; +} + +static int snd_msnd_playback_close(struct snd_pcm_substream *substream) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + snd_msnd_disable_irq(chip); + clear_bit(F_AUDIO_WRITE_INUSE, &chip->flags); + return 0; +} + + +static int snd_msnd_playback_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int i; + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + void *pDAQ = chip->mappedbase + DAPQ_DATA_BUFF; + + chip->play_sample_size = snd_pcm_format_width(params_format(params)); + chip->play_channels = params_channels(params); + chip->play_sample_rate = params_rate(params); + + for (i = 0; i < 3; ++i, pDAQ += DAQDS__size) { + writew(chip->play_sample_size, pDAQ + DAQDS_wSampleSize); + writew(chip->play_channels, pDAQ + DAQDS_wChannels); + writew(chip->play_sample_rate, pDAQ + DAQDS_wSampleRate); + } + /* dont do this here: + * snd_msnd_calibrate_adc(chip->play_sample_rate); + */ + + return 0; +} + +static int snd_msnd_playback_prepare(struct snd_pcm_substream *substream) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + unsigned int pcm_size = snd_pcm_lib_buffer_bytes(substream); + unsigned int pcm_count = snd_pcm_lib_period_bytes(substream); + unsigned int pcm_periods = pcm_size / pcm_count; + + snd_msnd_play_reset_queue(chip, pcm_periods, pcm_count); + chip->playDMAPos = 0; + return 0; +} + +static int snd_msnd_playback_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + int result = 0; + + if (cmd == SNDRV_PCM_TRIGGER_START) { + snd_printdd("snd_msnd_playback_trigger(START)\n"); + chip->banksPlayed = 0; + set_bit(F_WRITING, &chip->flags); + snd_msnd_DAPQ(chip, 1); + } else if (cmd == SNDRV_PCM_TRIGGER_STOP) { + snd_printdd("snd_msnd_playback_trigger(STop)\n"); + /* interrupt diagnostic, comment this out later */ + clear_bit(F_WRITING, &chip->flags); + snd_msnd_send_dsp_cmd(chip, HDEX_PLAY_STOP); + } else { + snd_printd(KERN_ERR "snd_msnd_playback_trigger(?????)\n"); + result = -EINVAL; + } + + snd_printdd("snd_msnd_playback_trigger() ENDE\n"); + return result; +} + +static snd_pcm_uframes_t +snd_msnd_playback_pointer(struct snd_pcm_substream *substream) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + return bytes_to_frames(substream->runtime, chip->playDMAPos); +} + + +static struct snd_pcm_ops snd_msnd_playback_ops = { + .open = snd_msnd_playback_open, + .close = snd_msnd_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_msnd_playback_hw_params, + .prepare = snd_msnd_playback_prepare, + .trigger = snd_msnd_playback_trigger, + .pointer = snd_msnd_playback_pointer, +}; + +static int snd_msnd_capture_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + set_bit(F_AUDIO_READ_INUSE, &chip->flags); + snd_msnd_enable_irq(chip); + runtime->dma_area = chip->mappedbase + 0x3000; + runtime->dma_bytes = 0x3000; + memset(runtime->dma_area, 0, runtime->dma_bytes); + chip->capture_substream = substream; + runtime->hw = snd_msnd_capture; + return 0; +} + +static int snd_msnd_capture_close(struct snd_pcm_substream *substream) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + snd_msnd_disable_irq(chip); + clear_bit(F_AUDIO_READ_INUSE, &chip->flags); + return 0; +} + +static int snd_msnd_capture_prepare(struct snd_pcm_substream *substream) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + unsigned int pcm_size = snd_pcm_lib_buffer_bytes(substream); + unsigned int pcm_count = snd_pcm_lib_period_bytes(substream); + unsigned int pcm_periods = pcm_size / pcm_count; + + snd_msnd_capture_reset_queue(chip, pcm_periods, pcm_count); + chip->captureDMAPos = 0; + return 0; +} + +static int snd_msnd_capture_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + if (cmd == SNDRV_PCM_TRIGGER_START) { + chip->last_recbank = -1; + set_bit(F_READING, &chip->flags); + if (snd_msnd_send_dsp_cmd(chip, HDEX_RECORD_START) == 0) + return 0; + + clear_bit(F_READING, &chip->flags); + } else if (cmd == SNDRV_PCM_TRIGGER_STOP) { + clear_bit(F_READING, &chip->flags); + snd_msnd_send_dsp_cmd(chip, HDEX_RECORD_STOP); + return 0; + } + return -EINVAL; +} + + +static snd_pcm_uframes_t +snd_msnd_capture_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + return bytes_to_frames(runtime, chip->captureDMAPos); +} + + +static int snd_msnd_capture_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int i; + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + void *pDAQ = chip->mappedbase + DARQ_DATA_BUFF; + + chip->capture_sample_size = snd_pcm_format_width(params_format(params)); + chip->capture_channels = params_channels(params); + chip->capture_sample_rate = params_rate(params); + + for (i = 0; i < 3; ++i, pDAQ += DAQDS__size) { + writew(chip->capture_sample_size, pDAQ + DAQDS_wSampleSize); + writew(chip->capture_channels, pDAQ + DAQDS_wChannels); + writew(chip->capture_sample_rate, pDAQ + DAQDS_wSampleRate); + } + return 0; +} + + +static struct snd_pcm_ops snd_msnd_capture_ops = { + .open = snd_msnd_capture_open, + .close = snd_msnd_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_msnd_capture_hw_params, + .prepare = snd_msnd_capture_prepare, + .trigger = snd_msnd_capture_trigger, + .pointer = snd_msnd_capture_pointer, +}; + + +int snd_msnd_pcm(struct snd_card *card, int device, + struct snd_pcm **rpcm) +{ + struct snd_msnd *chip = card->private_data; + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(card, "MSNDPINNACLE", device, 1, 1, &pcm); + if (err < 0) + return err; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_msnd_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_msnd_capture_ops); + + pcm->private_data = chip; + strcpy(pcm->name, "Hurricane"); + + + if (rpcm) + *rpcm = pcm; + return 0; +} +EXPORT_SYMBOL(snd_msnd_pcm); diff -urpN linux-rc5/sound/isa/msnd/msnd.h linux-git/sound/isa/msnd/msnd.h --- linux-rc5/sound/isa/msnd/msnd.h 1970-01-01 01:00:00.000000000 +0100 +++ linux-git/sound/isa/msnd/msnd.h 2009-01-22 21:17:09.125632934 +0100 @@ -0,0 +1,287 @@ +/********************************************************************* + * + * msnd.h + * + * Turtle Beach MultiSound Sound Card Driver for Linux + * + * Some parts of this header file were derived from the Turtle Beach + * MultiSound Driver Development Kit. + * + * Copyright (C) 1998 Andrew Veliath + * Copyright (C) 1993 Turtle Beach Systems, Inc. + * + * 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., 675 Mass Ave, Cambridge, MA 02139, USA. + * + ********************************************************************/ +#ifndef __MSND_H +#define __MSND_H + +#define DEFSAMPLERATE 44100 +#define DEFSAMPLESIZE SNDRV_PCM_FORMAT_S16 +#define DEFCHANNELS 1 + +#define SRAM_BANK_SIZE 0x8000 +#define SRAM_CNTL_START 0x7F00 +#define SMA_STRUCT_START 0x7F40 + +#define DSP_BASE_ADDR 0x4000 +#define DSP_BANK_BASE 0x4000 + +#define AGND 0x01 +#define SIGNAL 0x02 + +#define EXT_DSP_BIT_DCAL 0x0001 +#define EXT_DSP_BIT_MIDI_CON 0x0002 + +#define BUFFSIZE 0x8000 +#define HOSTQ_SIZE 0x40 + +#define DAP_BUFF_SIZE 0x2400 + +#define DAPQ_STRUCT_SIZE 0x10 +#define DARQ_STRUCT_SIZE 0x10 +#define DAPQ_BUFF_SIZE (3 * 0x10) +#define DARQ_BUFF_SIZE (3 * 0x10) +#define MODQ_BUFF_SIZE 0x400 + +#define DAPQ_DATA_BUFF 0x6C00 +#define DARQ_DATA_BUFF 0x6C30 +#define MODQ_DATA_BUFF 0x6C60 +#define MIDQ_DATA_BUFF 0x7060 + +#define DAPQ_OFFSET SRAM_CNTL_START +#define DARQ_OFFSET (SRAM_CNTL_START + 0x08) +#define MODQ_OFFSET (SRAM_CNTL_START + 0x10) +#define MIDQ_OFFSET (SRAM_CNTL_START + 0x18) +#define DSPQ_OFFSET (SRAM_CNTL_START + 0x20) + +#define HP_ICR 0x00 +#define HP_CVR 0x01 +#define HP_ISR 0x02 +#define HP_IVR 0x03 +#define HP_NU 0x04 +#define HP_INFO 0x04 +#define HP_TXH 0x05 +#define HP_RXH 0x05 +#define HP_TXM 0x06 +#define HP_RXM 0x06 +#define HP_TXL 0x07 +#define HP_RXL 0x07 + +#define HP_ICR_DEF 0x00 +#define HP_CVR_DEF 0x12 +#define HP_ISR_DEF 0x06 +#define HP_IVR_DEF 0x0f +#define HP_NU_DEF 0x00 + +#define HP_IRQM 0x09 + +#define HPR_BLRC 0x08 +#define HPR_SPR1 0x09 +#define HPR_SPR2 0x0A +#define HPR_TCL0 0x0B +#define HPR_TCL1 0x0C +#define HPR_TCL2 0x0D +#define HPR_TCL3 0x0E +#define HPR_TCL4 0x0F + +#define HPICR_INIT 0x80 +#define HPICR_HM1 0x40 +#define HPICR_HM0 0x20 +#define HPICR_HF1 0x10 +#define HPICR_HF0 0x08 +#define HPICR_TREQ 0x02 +#define HPICR_RREQ 0x01 + +#define HPCVR_HC 0x80 + +#define HPISR_HREQ 0x80 +#define HPISR_DMA 0x40 +#define HPISR_HF3 0x10 +#define HPISR_HF2 0x08 +#define HPISR_TRDY 0x04 +#define HPISR_TXDE 0x02 +#define HPISR_RXDF 0x01 + +#define HPIO_290 0 +#define HPIO_260 1 +#define HPIO_250 2 +#define HPIO_240 3 +#define HPIO_230 4 +#define HPIO_220 5 +#define HPIO_210 6 +#define HPIO_3E0 7 + +#define HPMEM_NONE 0 +#define HPMEM_B000 1 +#define HPMEM_C800 2 +#define HPMEM_D000 3 +#define HPMEM_D400 4 +#define HPMEM_D800 5 +#define HPMEM_E000 6 +#define HPMEM_E800 7 + +#define HPIRQ_NONE 0 +#define HPIRQ_5 1 +#define HPIRQ_7 2 +#define HPIRQ_9 3 +#define HPIRQ_10 4 +#define HPIRQ_11 5 +#define HPIRQ_12 6 +#define HPIRQ_15 7 + +#define HIMT_PLAY_DONE 0x00 +#define HIMT_RECORD_DONE 0x01 +#define HIMT_MIDI_EOS 0x02 +#define HIMT_MIDI_OUT 0x03 + +#define HIMT_MIDI_IN_UCHAR 0x0E +#define HIMT_DSP 0x0F + +#define HDEX_BASE 0x92 +#define HDEX_PLAY_START (0 + HDEX_BASE) +#define HDEX_PLAY_STOP (1 + HDEX_BASE) +#define HDEX_PLAY_PAUSE (2 + HDEX_BASE) +#define HDEX_PLAY_RESUME (3 + HDEX_BASE) +#define HDEX_RECORD_START (4 + HDEX_BASE) +#define HDEX_RECORD_STOP (5 + HDEX_BASE) +#define HDEX_MIDI_IN_START (6 + HDEX_BASE) +#define HDEX_MIDI_IN_STOP (7 + HDEX_BASE) +#define HDEX_MIDI_OUT_START (8 + HDEX_BASE) +#define HDEX_MIDI_OUT_STOP (9 + HDEX_BASE) +#define HDEX_AUX_REQ (10 + HDEX_BASE) + +#define HIWORD(l) ((u16)((((u32)(l)) >> 16) & 0xFFFF)) +#define LOWORD(l) ((u16)(u32)(l)) +#define HIBYTE(w) ((u8)(((u16)(w) >> 8) & 0xFF)) +#define LOBYTE(w) ((u8)(w)) +#define MAKELONG(low, hi) ((long)(((u16)(low))|(((u32)((u16)(hi)))<<16))) +#define MAKEWORD(low, hi) ((u16)(((u8)(low))|(((u16)((u8)(hi)))<<8))) + +#define PCTODSP_OFFSET(w) (u16)((w)/2) +#define PCTODSP_BASED(w) (u16)(((w)/2) + DSP_BASE_ADDR) +#define DSPTOPC_BASED(w) (((w) - DSP_BASE_ADDR) * 2) + +#ifdef SLOWIO +# undef outb +# undef inb +# define outb outb_p +# define inb inb_p +#endif + +/* JobQueueStruct */ +#define JQS_wStart 0x00 +#define JQS_wSize 0x02 +#define JQS_wHead 0x04 +#define JQS_wTail 0x06 +#define JQS__size 0x08 + +/* DAQueueDataStruct */ +#define DAQDS_wStart 0x00 +#define DAQDS_wSize 0x02 +#define DAQDS_wFormat 0x04 +#define DAQDS_wSampleSize 0x06 +#define DAQDS_wChannels 0x08 +#define DAQDS_wSampleRate 0x0A +#define DAQDS_wIntMsg 0x0C +#define DAQDS_wFlags 0x0E +#define DAQDS__size 0x10 + +struct snd_msnd { + void __iomem *mappedbase; + int play_period_bytes; + int playLimit; + int playPeriods; + int playDMAPos; + int banksPlayed; + int captureDMAPos; + int capturePeriodBytes; + int captureLimit; + int capturePeriods; + struct snd_card *card; + void *msndmidi_mpu; + struct snd_rawmidi *rmidi; + + /* Hardware resources */ + long io; + int memid, irqid; + int irq, irq_ref; + unsigned long base; + + /* Motorola 56k DSP SMA */ + void __iomem *SMA; + void __iomem *DAPQ; + void __iomem *DARQ; + void __iomem *MODQ; + void __iomem *MIDQ; + void __iomem *DSPQ; + int dspq_data_buff, dspq_buff_size; + + /* State variables */ + enum { msndClassic, msndPinnacle } type; + mode_t mode; + unsigned long flags; +#define F_RESETTING 0 +#define F_HAVEDIGITAL 1 +#define F_AUDIO_WRITE_INUSE 2 +#define F_WRITING 3 +#define F_WRITEBLOCK 4 +#define F_WRITEFLUSH 5 +#define F_AUDIO_READ_INUSE 6 +#define F_READING 7 +#define F_READBLOCK 8 +#define F_EXT_MIDI_INUSE 9 +#define F_HDR_MIDI_INUSE 10 +#define F_DISABLE_WRITE_NDELAY 11 + spinlock_t lock; + spinlock_t mixer_lock; + int nresets; + unsigned recsrc; +#define LEVEL_ENTRIES 32 + int left_levels[LEVEL_ENTRIES]; + int right_levels[LEVEL_ENTRIES]; + int calibrate_signal; + int play_sample_size, play_sample_rate, play_channels; + int play_ndelay; + int capture_sample_size, capture_sample_rate, capture_channels; + int capture_ndelay; + u8 bCurrentMidiPatch; + + int last_playbank, last_recbank; + struct snd_pcm_substream *playback_substream; + struct snd_pcm_substream *capture_substream; + +}; + +void snd_msnd_init_queue(void *base, int start, int size); + +int snd_msnd_send_dsp_cmd(struct snd_msnd *chip, u8 cmd); +int snd_msnd_send_word(struct snd_msnd *chip, + unsigned char high, + unsigned char mid, + unsigned char low); +int snd_msnd_upload_host(struct snd_msnd *chip, + const u8 *bin, int len); +int snd_msnd_enable_irq(struct snd_msnd *chip); +int snd_msnd_disable_irq(struct snd_msnd *chip); +void snd_msnd_dsp_halt(struct snd_msnd *chip, struct file *file); +int snd_msnd_DAPQ(struct snd_msnd *chip, int start); +int snd_msnd_DARQ(struct snd_msnd *chip, int start); +int snd_msnd_pcm(struct snd_card *card, int device, struct snd_pcm **rpcm); + +int snd_msndmidi_new(struct snd_card *card, int device); +void snd_msndmidi_input_read(void *mpu); + +#endif /* __MSND_H */ diff -urpN linux-rc5/sound/isa/msnd/msnd_classic.c linux-git/sound/isa/msnd/msnd_classic.c --- linux-rc5/sound/isa/msnd/msnd_classic.c 1970-01-01 01:00:00.000000000 +0100 +++ linux-git/sound/isa/msnd/msnd_classic.c 2009-01-16 22:00:53.058024371 +0100 @@ -0,0 +1,3 @@ +/* The work is in msnd_pinnacle.c, just define MSND_CLASSIC before it. */ +#define MSND_CLASSIC +#include "msnd_pinnacle.c" diff -urpN linux-rc5/sound/isa/msnd/msnd_classic.h linux-git/sound/isa/msnd/msnd_classic.h --- linux-rc5/sound/isa/msnd/msnd_classic.h 1970-01-01 01:00:00.000000000 +0100 +++ linux-git/sound/isa/msnd/msnd_classic.h 2009-01-22 21:03:28.055334713 +0100 @@ -0,0 +1,140 @@ +/********************************************************************* + * + * msnd_classic.h + * + * Turtle Beach MultiSound Sound Card Driver for Linux + * + * Some parts of this header file were derived from the Turtle Beach + * MultiSound Driver Development Kit. + * + * Copyright (C) 1998 Andrew Veliath + * Copyright (C) 1993 Turtle Beach Systems, Inc. + * + * 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., 675 Mass Ave, Cambridge, MA 02139, USA. + * + ********************************************************************/ +#ifndef __MSND_CLASSIC_H +#define __MSND_CLASSIC_H + +#define DSP_NUMIO 0x10 + +#define HP_MEMM 0x08 + +#define HP_BITM 0x0E +#define HP_WAIT 0x0D +#define HP_DSPR 0x0A +#define HP_PROR 0x0B +#define HP_BLKS 0x0C + +#define HPPRORESET_OFF 0 +#define HPPRORESET_ON 1 + +#define HPDSPRESET_OFF 0 +#define HPDSPRESET_ON 1 + +#define HPBLKSEL_0 0 +#define HPBLKSEL_1 1 + +#define HPWAITSTATE_0 0 +#define HPWAITSTATE_1 1 + +#define HPBITMODE_16 0 +#define HPBITMODE_8 1 + +#define HIDSP_INT_PLAY_UNDER 0x00 +#define HIDSP_INT_RECORD_OVER 0x01 +#define HIDSP_INPUT_CLIPPING 0x02 +#define HIDSP_MIDI_IN_OVER 0x10 +#define HIDSP_MIDI_OVERRUN_ERR 0x13 + +#define HDEXAR_CLEAR_PEAKS 1 +#define HDEXAR_IN_SET_POTS 2 +#define HDEXAR_AUX_SET_POTS 3 +#define HDEXAR_CAL_A_TO_D 4 +#define HDEXAR_RD_EXT_DSP_BITS 5 + +#define TIME_PRO_RESET_DONE 0x028A +#define TIME_PRO_SYSEX 0x0040 +#define TIME_PRO_RESET 0x0032 + +#define DAR_BUFF_SIZE 0x2000 + +#define MIDQ_BUFF_SIZE 0x200 +#define DSPQ_BUFF_SIZE 0x40 + +#define DSPQ_DATA_BUFF 0x7260 + +#define MOP_SYNTH 0x10 +#define MOP_EXTOUT 0x32 +#define MOP_EXTTHRU 0x02 +#define MOP_OUTMASK 0x01 + +#define MIP_EXTIN 0x01 +#define MIP_SYNTH 0x00 +#define MIP_INMASK 0x32 + +/* Classic SMA Common Data */ +#define SMA_wCurrPlayBytes 0x0000 +#define SMA_wCurrRecordBytes 0x0002 +#define SMA_wCurrPlayVolLeft 0x0004 +#define SMA_wCurrPlayVolRight 0x0006 +#define SMA_wCurrInVolLeft 0x0008 +#define SMA_wCurrInVolRight 0x000a +#define SMA_wUser_3 0x000c +#define SMA_wUser_4 0x000e +#define SMA_dwUser_5 0x0010 +#define SMA_dwUser_6 0x0014 +#define SMA_wUser_7 0x0018 +#define SMA_wReserved_A 0x001a +#define SMA_wReserved_B 0x001c +#define SMA_wReserved_C 0x001e +#define SMA_wReserved_D 0x0020 +#define SMA_wReserved_E 0x0022 +#define SMA_wReserved_F 0x0024 +#define SMA_wReserved_G 0x0026 +#define SMA_wReserved_H 0x0028 +#define SMA_wCurrDSPStatusFlags 0x002a +#define SMA_wCurrHostStatusFlags 0x002c +#define SMA_wCurrInputTagBits 0x002e +#define SMA_wCurrLeftPeak 0x0030 +#define SMA_wCurrRightPeak 0x0032 +#define SMA_wExtDSPbits 0x0034 +#define SMA_bExtHostbits 0x0036 +#define SMA_bBoardLevel 0x0037 +#define SMA_bInPotPosRight 0x0038 +#define SMA_bInPotPosLeft 0x0039 +#define SMA_bAuxPotPosRight 0x003a +#define SMA_bAuxPotPosLeft 0x003b +#define SMA_wCurrMastVolLeft 0x003c +#define SMA_wCurrMastVolRight 0x003e +#define SMA_bUser_12 0x0040 +#define SMA_bUser_13 0x0041 +#define SMA_wUser_14 0x0042 +#define SMA_wUser_15 0x0044 +#define SMA_wCalFreqAtoD 0x0046 +#define SMA_wUser_16 0x0048 +#define SMA_wUser_17 0x004a +#define SMA__size 0x004c + +#define INITCODEFILE "turtlebeach/msndinit.bin" +#define PERMCODEFILE "turtlebeach/msndperm.bin" +#define LONGNAME "MultiSound (Classic/Monterey/Tahiti)" + +/* functions from external mixer file */ +void snd_msndmix_setup(struct snd_msnd *chip); +unsigned long snd_msndmix_force_recsrc(struct snd_msnd *chip, int recsrc); +int __devinit snd_msndmix_new(struct snd_card *card); + +#endif /* __MSND_CLASSIC_H */ diff -urpN linux-rc5/sound/isa/msnd/msnd_midi.c linux-git/sound/isa/msnd/msnd_midi.c --- linux-rc5/sound/isa/msnd/msnd_midi.c 1970-01-01 01:00:00.000000000 +0100 +++ linux-git/sound/isa/msnd/msnd_midi.c 2009-01-22 18:35:00.942072608 +0100 @@ -0,0 +1,179 @@ +/* + * Copyright (c) by Jaroslav Kysela <perex(a)perex.cz> + * Routines for control of MPU-401 in UART mode + * + * MPU-401 supports UART mode which is not capable generate transmit + * interrupts thus output is done via polling. Also, if irq < 0, then + * input is done also via polling. Do not expect good performance. + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <linux/io.h> +#include <linux/delay.h> +#include <linux/ioport.h> +#include <linux/errno.h> +#include <sound/core.h> +#include <sound/rawmidi.h> + +#include "msnd.h" + +#define MSNDMIDI_MODE_BIT_INPUT 0 +#define MSNDMIDI_MODE_BIT_OUTPUT 1 +#define MSNDMIDI_MODE_BIT_INPUT_TRIGGER 2 +#define MSNDMIDI_MODE_BIT_OUTPUT_TRIGGER 3 + +struct snd_msndmidi { + struct snd_msnd *dev; + + unsigned long mode; /* MSNDMIDI_MODE_XXXX */ + + struct snd_rawmidi_substream *substream_input; + + spinlock_t input_lock; +}; + +/* + * input/output open/close - protected by open_mutex in rawmidi.c + */ +static int snd_msndmidi_input_open(struct snd_rawmidi_substream *substream) +{ + struct snd_msndmidi *mpu; + + snd_printdd("snd_msndmidi_input_open()\n"); + + mpu = substream->rmidi->private_data; + + mpu->substream_input = substream; + + snd_msnd_enable_irq(mpu->dev); + + snd_msnd_send_dsp_cmd(mpu->dev, HDEX_MIDI_IN_START); + set_bit(MSNDMIDI_MODE_BIT_INPUT, &mpu->mode); + return 0; +} + +static int snd_msndmidi_input_close(struct snd_rawmidi_substream *substream) +{ + struct snd_msndmidi *mpu; + + mpu = substream->rmidi->private_data; + snd_msnd_send_dsp_cmd(mpu->dev, HDEX_MIDI_IN_STOP); + clear_bit(MSNDMIDI_MODE_BIT_INPUT, &mpu->mode); + mpu->substream_input = NULL; + snd_msnd_disable_irq(mpu->dev); + return 0; +} + +static void snd_msndmidi_input_drop(struct snd_msndmidi *mpu) +{ + u16 tail; + + tail = readw(mpu->dev->MIDQ + JQS_wTail); + writew(tail, mpu->dev->MIDQ + JQS_wHead); +} + +/* + * trigger input + */ +static void snd_msndmidi_input_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + unsigned long flags; + struct snd_msndmidi *mpu; + + snd_printdd("snd_msndmidi_input_trigger(, %i)\n", up); + + mpu = substream->rmidi->private_data; + spin_lock_irqsave(&mpu->input_lock, flags); + if (up) { + if (!test_and_set_bit(MSNDMIDI_MODE_BIT_INPUT_TRIGGER, + &mpu->mode)) + snd_msndmidi_input_drop(mpu); + } else { + clear_bit(MSNDMIDI_MODE_BIT_INPUT_TRIGGER, &mpu->mode); + } + spin_unlock_irqrestore(&mpu->input_lock, flags); + if (up) + snd_msndmidi_input_read(mpu); +} + +void snd_msndmidi_input_read(void *mpuv) +{ + unsigned long flags; + struct snd_msndmidi *mpu = mpuv; + void *pwMIDQData = mpu->dev->mappedbase + MIDQ_DATA_BUFF; + + spin_lock_irqsave(&mpu->input_lock, flags); + while (readw(mpu->dev->MIDQ + JQS_wTail) != + readw(mpu->dev->MIDQ + JQS_wHead)) { + u16 wTmp, val; + val = readw(pwMIDQData + 2 * readw(mpu->dev->MIDQ + JQS_wHead)); + + if (test_bit(MSNDMIDI_MODE_BIT_INPUT_TRIGGER, + &mpu->mode)) + snd_rawmidi_receive(mpu->substream_input, + (unsigned char *)&val, 1); + + wTmp = readw(mpu->dev->MIDQ + JQS_wHead) + 1; + if (wTmp > readw(mpu->dev->MIDQ + JQS_wSize)) + writew(0, mpu->dev->MIDQ + JQS_wHead); + else + writew(wTmp, mpu->dev->MIDQ + JQS_wHead); + } + spin_unlock_irqrestore(&mpu->input_lock, flags); +} +EXPORT_SYMBOL(snd_msndmidi_input_read); + +static struct snd_rawmidi_ops snd_msndmidi_input = { + .open = snd_msndmidi_input_open, + .close = snd_msndmidi_input_close, + .trigger = snd_msndmidi_input_trigger, +}; + +static void snd_msndmidi_free(struct snd_rawmidi *rmidi) +{ + struct snd_msndmidi *mpu = rmidi->private_data; + kfree(mpu); +} + +int snd_msndmidi_new(struct snd_card *card, int device) +{ + struct snd_msnd *chip = card->private_data; + struct snd_msndmidi *mpu; + struct snd_rawmidi *rmidi; + int err; + + err = snd_rawmidi_new(card, "MSND-MIDI", device, 1, 1, &rmidi); + if (err < 0) + return err; + mpu = kcalloc(1, sizeof(*mpu), GFP_KERNEL); + if (mpu == NULL) { + snd_device_free(card, rmidi); + return -ENOMEM; + } + mpu->dev = chip; + chip->msndmidi_mpu = mpu; + rmidi->private_data = mpu; + rmidi->private_free = snd_msndmidi_free; + spin_lock_init(&mpu->input_lock); + strcpy(rmidi->name, "MSND MIDI"); + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &snd_msndmidi_input); + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; + return 0; +} diff -urpN linux-rc5/sound/isa/msnd/msnd_pinnacle.c linux-git/sound/isa/msnd/msnd_pinnacle.c --- linux-rc5/sound/isa/msnd/msnd_pinnacle.c 1970-01-01 01:00:00.000000000 +0100 +++ linux-git/sound/isa/msnd/msnd_pinnacle.c 2009-01-22 21:19:31.452624274 +0100 @@ -0,0 +1,1244 @@ +/********************************************************************* + * + * Linux multisound pinnacle/fiji driver for ALSA. + * + * 2002/06/30 Karsten Wiese: + * for now this is only used to build a pinnacle / fiji driver. + * the OSS parent of this code is designed to also support + * the multisound classic via the file msnd_classic.c. + * to make it easier for some brave heart to implemt classic + * support in alsa, i left all the MSND_CLASSIC tokens in this file. + * but for now this untested & undone. + * + * + * ripped from linux kernel 2.4.18 by Karsten Wiese. + * + * the following is a copy of the 2.4.18 OSS FREE file-heading comment: + * + * Turtle Beach MultiSound Sound Card Driver for Linux + * msnd_pinnacle.c / msnd_classic.c + * + * -- If MSND_CLASSIC is defined: + * + * -> driver for Turtle Beach Classic/Monterey/Tahiti + * + * -- Else + * + * -> driver for Turtle Beach Pinnacle/Fiji + * + * 12-3-2000 Modified IO port validation Steve Sycamore + * + * Copyright (C) 1998 Andrew Veliath + * + * 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., 675 Mass Ave, Cambridge, MA 02139, USA. + * + ********************************************************************/ + +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/types.h> +#include <linux/delay.h> +#include <linux/ioport.h> +#include <linux/firmware.h> +#include <linux/isa.h> +#include <linux/isapnp.h> +#include <linux/irq.h> +#include <linux/io.h> + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/asound.h> +#include <sound/pcm.h> +#include <sound/mpu401.h> + +#ifdef MSND_CLASSIC +# ifndef __alpha__ +# define SLOWIO +# endif +#endif +#include "msnd.h" +#ifdef MSND_CLASSIC +# include "msnd_classic.h" +# define LOGNAME "msnd_classic" +#else +# include "msnd_pinnacle.h" +# define LOGNAME "snd_msnd_pinnacle" +#endif + +static void set_default_audio_parameters(struct snd_msnd *chip) +{ + chip->play_sample_size = DEFSAMPLESIZE; + chip->play_sample_rate = DEFSAMPLERATE; + chip->play_channels = DEFCHANNELS; + chip->capture_sample_size = DEFSAMPLESIZE; + chip->capture_sample_rate = DEFSAMPLERATE; + chip->capture_channels = DEFCHANNELS; +} + +static void snd_msnd_eval_dsp_msg(struct snd_msnd *chip, u16 wMessage) +{ + switch (HIBYTE(wMessage)) { + case HIMT_PLAY_DONE: { + #ifdef CONFIG_SND_DEBUG0 + if (chip->banksPlayed < 3) + snd_printdd("%08X: HIMT_PLAY_DONE: %i\n", + (unsigned)jiffies, LOBYTE(wMessage)); + #endif + + if (chip->last_playbank == LOBYTE(wMessage)) { + snd_printdd("chip.last_playbank == LOBYTE(wMessage)\n"); + break; + } + chip->banksPlayed++; + + if (test_bit(F_WRITING, &chip->flags)) + snd_msnd_DAPQ(chip, 0); + + chip->last_playbank = LOBYTE(wMessage); + chip->playDMAPos += chip->play_period_bytes; + if (chip->playDMAPos > chip->playLimit) + chip->playDMAPos = 0; + snd_pcm_period_elapsed(chip->playback_substream); + + break; + } + case HIMT_RECORD_DONE: + if (chip->last_recbank == LOBYTE(wMessage)) + break; + chip->last_recbank = LOBYTE(wMessage); + chip->captureDMAPos += chip->capturePeriodBytes; + if (chip->captureDMAPos > (chip->captureLimit)) + chip->captureDMAPos = 0; + + if (test_bit(F_READING, &chip->flags)) + snd_msnd_DARQ(chip, chip->last_recbank); + + snd_pcm_period_elapsed(chip->capture_substream); + break; + + case HIMT_DSP: + switch (LOBYTE(wMessage)) { +#ifndef MSND_CLASSIC + case HIDSP_PLAY_UNDER: +#endif + case HIDSP_INT_PLAY_UNDER: + snd_printd(KERN_WARNING LOGNAME ": Play underflow %i\n", + chip->banksPlayed); + if (chip->banksPlayed > 2) + clear_bit(F_WRITING, &chip->flags); + break; + + case HIDSP_INT_RECORD_OVER: + snd_printd(KERN_WARNING LOGNAME ": Record overflow\n"); + clear_bit(F_READING, &chip->flags); + break; + + default: + snd_printd(KERN_WARNING LOGNAME + ": DSP message %d 0x%02x\n", + LOBYTE(wMessage), LOBYTE(wMessage)); + break; + } + break; + + case HIMT_MIDI_IN_UCHAR: + if (chip->msndmidi_mpu) + snd_msndmidi_input_read(chip->msndmidi_mpu); + break; + + default: + snd_printd(KERN_WARNING LOGNAME ": HIMT message %d 0x%02x\n", + HIBYTE(wMessage), HIBYTE(wMessage)); + break; + } +} + +static irqreturn_t snd_msnd_interrupt(int irq, void *dev_id) +{ + struct snd_msnd *chip = dev_id; + void *pwDSPQData = chip->mappedbase + DSPQ_DATA_BUFF; + + /* Send ack to DSP */ + /* inb(chip->io + HP_RXL); */ + + /* Evaluate queued DSP messages */ + while (readw(chip->DSPQ + JQS_wTail) != readw(chip->DSPQ + JQS_wHead)) { + u16 wTmp; + + snd_msnd_eval_dsp_msg(chip, + readw(pwDSPQData + 2 * readw(chip->DSPQ + JQS_wHead))); + + wTmp = readw(chip->DSPQ + JQS_wHead) + 1; + if (wTmp > readw(chip->DSPQ + JQS_wSize)) + writew(0, chip->DSPQ + JQS_wHead); + else + writew(wTmp, chip->DSPQ + JQS_wHead); + } + /* Send ack to DSP */ + inb(chip->io + HP_RXL); + return IRQ_HANDLED; +} + + +static int snd_msnd_reset_dsp(long io, unsigned char *info) +{ + int timeout = 100; + + outb(HPDSPRESET_ON, io + HP_DSPR); + msleep(1); +#ifndef MSND_CLASSIC + if (info) + *info = inb(io + HP_INFO); +#endif + outb(HPDSPRESET_OFF, io + HP_DSPR); + msleep(1); + while (timeout-- > 0) { + if (inb(io + HP_CVR) == HP_CVR_DEF) + return 0; + msleep(1); + } + snd_printk(KERN_ERR LOGNAME ": Cannot reset DSP\n"); + + return -EIO; +} + +static int __devinit snd_msnd_probe(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + unsigned char info; +#ifndef MSND_CLASSIC + char *xv, *rev = NULL; + char *pin = "TB Pinnacle", *fiji = "TB Fiji"; + char *pinfiji = "TB Pinnacle/Fiji"; +#endif + + if (!request_region(chip->io, DSP_NUMIO, "probing")) { + snd_printk(KERN_ERR LOGNAME ": I/O port conflict\n"); + return -ENODEV; + } + + if (snd_msnd_reset_dsp(chip->io, &info) < 0) { + release_region(chip->io, DSP_NUMIO); + return -ENODEV; + } + +#ifdef MSND_CLASSIC + strcpy(card->shortname, "Classic/Tahiti/Monterey"); + printk(KERN_INFO LOGNAME ": %s, " +#else + switch (info >> 4) { + case 0xf: + xv = "<= 1.15"; + break; + case 0x1: + xv = "1.18/1.2"; + break; + case 0x2: + xv = "1.3"; + break; + case 0x3: + xv = "1.4"; + break; + default: + xv = "unknown"; + break; + } + + switch (info & 0x7) { + case 0x0: + rev = "I"; + strcpy(card->shortname, pin); + break; + case 0x1: + rev = "F"; + strcpy(card->shortname, pin); + break; + case 0x2: + rev = "G"; + strcpy(card->shortname, pin); + break; + case 0x3: + rev = "H"; + strcpy(card->shortname, pin); + break; + case 0x4: + rev = "E"; + strcpy(card->shortname, fiji); + break; + case 0x5: + rev = "C"; + strcpy(card->shortname, fiji); + break; + case 0x6: + rev = "D"; + strcpy(card->shortname, fiji); + break; + case 0x7: + rev = "A-B (Fiji) or A-E (Pinnacle)"; + strcpy(card->shortname, pinfiji); + break; + } + printk(KERN_INFO LOGNAME ": %s revision %s, Xilinx version %s, " +#endif /* MSND_CLASSIC */ + "I/O 0x%lx-0x%lx, IRQ %d, memory mapped to 0x%lX-0x%lX\n", + card->shortname, +#ifndef MSND_CLASSIC + rev, xv, +#endif + chip->io, chip->io + DSP_NUMIO - 1, + chip->irq, + chip->base, chip->base + 0x7fff); + + strcpy(card->longname, "Turtle Beach Multisound Pinnacle"); + release_region(chip->io, DSP_NUMIO); + return 0; +} + +static int snd_msnd_init_sma(struct snd_msnd *chip) +{ + static int initted; + u16 mastVolLeft, mastVolRight; + unsigned long flags; + +#ifdef MSND_CLASSIC + outb(chip->memid, chip->io + HP_MEMM); +#endif + outb(HPBLKSEL_0, chip->io + HP_BLKS); + /* Motorola 56k shared memory base */ + chip->SMA = chip->mappedbase + SMA_STRUCT_START; + + if (initted) { + mastVolLeft = readw(chip->SMA + SMA_wCurrMastVolLeft); + mastVolRight = readw(chip->SMA + SMA_wCurrMastVolRight); + } else + mastVolLeft = mastVolRight = 0; + memset_io(chip->mappedbase, 0, 0x8000); + + /* Critical section: bank 1 access */ + spin_lock_irqsave(&chip->lock, flags); + outb(HPBLKSEL_1, chip->io + HP_BLKS); + memset_io(chip->mappedbase, 0, 0x8000); + outb(HPBLKSEL_0, chip->io + HP_BLKS); + spin_unlock_irqrestore(&chip->lock, flags); + + /* Digital audio play queue */ + chip->DAPQ = chip->mappedbase + DAPQ_OFFSET; + snd_msnd_init_queue(chip->DAPQ, DAPQ_DATA_BUFF, DAPQ_BUFF_SIZE); + + /* Digital audio record queue */ + chip->DARQ = chip->mappedbase + DARQ_OFFSET; + snd_msnd_init_queue(chip->DARQ, DARQ_DATA_BUFF, DARQ_BUFF_SIZE); + + /* MIDI out queue */ + chip->MODQ = chip->mappedbase + MODQ_OFFSET; + snd_msnd_init_queue(chip->MODQ, MODQ_DATA_BUFF, MODQ_BUFF_SIZE); + + /* MIDI in queue */ + chip->MIDQ = chip->mappedbase + MIDQ_OFFSET; + snd_msnd_init_queue(chip->MIDQ, MIDQ_DATA_BUFF, MIDQ_BUFF_SIZE); + + /* DSP -> host message queue */ + chip->DSPQ = chip->mappedbase + DSPQ_OFFSET; + snd_msnd_init_queue(chip->DSPQ, DSPQ_DATA_BUFF, DSPQ_BUFF_SIZE); + + /* Setup some DSP values */ +#ifndef MSND_CLASSIC + writew(1, chip->SMA + SMA_wCurrPlayFormat); + writew(chip->play_sample_size, chip->SMA + SMA_wCurrPlaySampleSize); + writew(chip->play_channels, chip->SMA + SMA_wCurrPlayChannels); + writew(chip->play_sample_rate, chip->SMA + SMA_wCurrPlaySampleRate); +#endif + writew(chip->play_sample_rate, chip->SMA + SMA_wCalFreqAtoD); + writew(mastVolLeft, chip->SMA + SMA_wCurrMastVolLeft); + writew(mastVolRight, chip->SMA + SMA_wCurrMastVolRight); +#ifndef MSND_CLASSIC + writel(0x00010000, chip->SMA + SMA_dwCurrPlayPitch); + writel(0x00000001, chip->SMA + SMA_dwCurrPlayRate); +#endif + writew(0x303, chip->SMA + SMA_wCurrInputTagBits); + + initted = 1; + + return 0; +} + + +static int upload_dsp_code(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + const struct firmware *init_fw = NULL, *perm_fw = NULL; + int err; + + outb(HPBLKSEL_0, chip->io + HP_BLKS); + + err = request_firmware(&init_fw, INITCODEFILE, card->dev); + if (err < 0) { + printk(KERN_ERR LOGNAME ": Error loading " INITCODEFILE); + goto cleanup1; + } + err = request_firmware(&perm_fw, PERMCODEFILE, card->dev); + if (err < 0) { + printk(KERN_ERR LOGNAME ": Error loading " PERMCODEFILE); + goto cleanup; + } + + memcpy_toio(chip->mappedbase, perm_fw->data, perm_fw->size); + if (snd_msnd_upload_host(chip, init_fw->data, init_fw->size) < 0) { + printk(KERN_WARNING LOGNAME ": Error uploading to DSP\n"); + err = -ENODEV; + goto cleanup; + } + printk(KERN_INFO LOGNAME ": DSP firmware uploaded\n"); + err = 0; + +cleanup: + release_firmware(perm_fw); +cleanup1: + release_firmware(init_fw); + return err; +} + +#ifdef MSND_CLASSIC +static void reset_proteus(struct snd_msnd *chip) +{ + outb(HPPRORESET_ON, chip->io + HP_PROR); + msleep(TIME_PRO_RESET); + outb(HPPRORESET_OFF, chip->io + HP_PROR); + msleep(TIME_PRO_RESET_DONE); +} +#endif + +static int snd_msnd_initialize(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + int err, timeout; + +#ifdef MSND_CLASSIC + outb(HPWAITSTATE_0, chip->io + HP_WAIT); + outb(HPBITMODE_16, chip->io + HP_BITM); + + reset_proteus(chip); +#endif + err = snd_msnd_init_sma(chip); + if (err < 0) { + printk(KERN_WARNING LOGNAME ": Cannot initialize SMA\n"); + return err; + } + + err = snd_msnd_reset_dsp(chip->io, NULL); + if (err < 0) + return err; + + err = upload_dsp_code(card); + if (err < 0) { + printk(KERN_WARNING LOGNAME ": Cannot upload DSP code\n"); + return err; + } + + timeout = 200; + + while (readw(chip->mappedbase)) { + msleep(1); + if (!timeout--) { + snd_printd(KERN_ERR LOGNAME ": DSP reset timeout\n"); + return -EIO; + } + } + + snd_msndmix_setup(chip); + return 0; +} + +static int snd_msnd_dsp_full_reset(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + int rv; + + if (test_bit(F_RESETTING, &chip->flags) || ++chip->nresets > 10) + return 0; + + set_bit(F_RESETTING, &chip->flags); + snd_msnd_dsp_halt(chip, NULL); /* Unconditionally halt */ + + rv = snd_msnd_initialize(card); + if (rv) + printk(KERN_WARNING LOGNAME ": DSP reset failed\n"); + snd_msndmix_force_recsrc(chip, 0); + clear_bit(F_RESETTING, &chip->flags); + return rv; +} + +static int snd_msnd_dev_free(struct snd_device *device) +{ + snd_printdd("snd_msnd_chip_free()\n"); + return 0; +} + +int snd_msnd_send_dsp_cmd_chk(struct snd_msnd *chip, u8 cmd) +{ + if (snd_msnd_send_dsp_cmd(chip, cmd) == 0) + return 0; + snd_msnd_dsp_full_reset(chip->card); + return snd_msnd_send_dsp_cmd(chip, cmd); +} + +static int __devinit snd_msnd_calibrate_adc(struct snd_msnd *chip, u16 srate) +{ + snd_printdd("snd_msnd_calibrate_adc(%i)\n", srate); + writew(srate, chip->SMA + SMA_wCalFreqAtoD); + if (chip->calibrate_signal == 0) + writew(readw(chip->SMA + SMA_wCurrHostStatusFlags) + | 0x0001, chip->SMA + SMA_wCurrHostStatusFlags); + else + writew(readw(chip->SMA + SMA_wCurrHostStatusFlags) + & ~0x0001, chip->SMA + SMA_wCurrHostStatusFlags); + if (snd_msnd_send_word(chip, 0, 0, HDEXAR_CAL_A_TO_D) == 0 && + snd_msnd_send_dsp_cmd_chk(chip, HDEX_AUX_REQ) == 0) { + schedule_timeout_interruptible(msecs_to_jiffies(333)); + return 0; + } + printk(KERN_WARNING LOGNAME ": ADC calibration failed\n"); + return -EIO; +} + +/* + * ALSA callback function, called when attempting to open the MIDI device. + */ +static int snd_msnd_mpu401_open(struct snd_mpu401 *mpu) +{ + snd_msnd_enable_irq(mpu->private_data); + snd_msnd_send_dsp_cmd(mpu->private_data, HDEX_MIDI_IN_START); + return 0; +} + +static void snd_msnd_mpu401_close(struct snd_mpu401 *mpu) +{ + snd_msnd_send_dsp_cmd(mpu->private_data, HDEX_MIDI_IN_STOP); + snd_msnd_disable_irq(mpu->private_data); +} + +static long mpu_io[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT; +static int mpu_irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ; + +static int __devinit snd_msnd_attach(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + int err; + static struct snd_device_ops ops = { + .dev_free = snd_msnd_dev_free, + }; + + err = request_irq(chip->irq, snd_msnd_interrupt, 0, card->shortname, + chip); + if (err < 0) { + printk(KERN_ERR LOGNAME ": Couldn't grab IRQ %d\n", chip->irq); + return err; + } + request_region(chip->io, DSP_NUMIO, card->shortname); + + if (!request_mem_region(chip->base, BUFFSIZE, card->shortname)) { + printk(KERN_ERR LOGNAME + ": unable to grab memory region 0x%lx-0x%lx\n", + chip->base, chip->base + BUFFSIZE - 1); + release_region(chip->io, DSP_NUMIO); + free_irq(chip->irq, chip); + return -EBUSY; + } + chip->mappedbase = ioremap(chip->base, 0x8000); + if (!chip->mappedbase) { + printk(KERN_ERR LOGNAME + ": unable to map memory region 0x%lx-0x%lx\n", + chip->base, chip->base + BUFFSIZE - 1); + err = -EIO; + goto err_release_region; + } + chip->mappedbase = ioremap(chip->base, 0x8000); + + err = snd_msnd_dsp_full_reset(card); + if (err < 0) + goto err_release_region; + + /* Register device */ + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) + goto err_release_region; + + err = snd_msnd_pcm(card, 0, NULL); + if (err < 0) { + printk(KERN_ERR LOGNAME ": error creating new PCM device\n"); + goto err_release_region; + } + + err = snd_msndmix_new(card); + if (err < 0) { + printk(KERN_ERR LOGNAME ": error creating new Mixer device\n"); + goto err_release_region; + } + + + if (mpu_io[0] != SNDRV_AUTO_PORT) { + struct snd_mpu401 *mpu; + + err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, + mpu_io[0], + MPU401_MODE_INPUT | + MPU401_MODE_OUTPUT, + mpu_irq[0], IRQF_DISABLED, + &chip->rmidi); + if (err < 0) { + printk(KERN_ERR LOGNAME + ": error creating new Midi device\n"); + goto err_release_region; + } + mpu = chip->rmidi->private_data; + + mpu->open_input = snd_msnd_mpu401_open; + mpu->close_input = snd_msnd_mpu401_close; + mpu->private_data = chip; + } + + disable_irq(chip->irq); + snd_msnd_calibrate_adc(chip, chip->play_sample_rate); + snd_msndmix_force_recsrc(chip, 0); + + err = snd_card_register(card); + if (err < 0) + goto err_release_region; + + return 0; + +err_release_region: + if (chip->mappedbase) + iounmap(chip->mappedbase); + release_mem_region(chip->base, BUFFSIZE); + release_region(chip->io, DSP_NUMIO); + free_irq(chip->irq, chip); + return err; +} + + +static void __devexit snd_msnd_unload(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + + iounmap(chip->mappedbase); + release_mem_region(chip->base, BUFFSIZE); + release_region(chip->io, DSP_NUMIO); + free_irq(chip->irq, chip); + snd_card_free(card); +} + +#ifndef MSND_CLASSIC + +/* Pinnacle/Fiji Logical Device Configuration */ + +static int __devinit snd_msnd_write_cfg(int cfg, int reg, int value) +{ + outb(reg, cfg); + outb(value, cfg + 1); + if (value != inb(cfg + 1)) { + printk(KERN_ERR LOGNAME ": snd_msnd_write_cfg: I/O error\n"); + return -EIO; + } + return 0; +} + +static int __devinit snd_msnd_write_cfg_io0(int cfg, int num, u16 io) +{ + if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_IO0_BASEHI, HIBYTE(io))) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_IO0_BASELO, LOBYTE(io))) + return -EIO; + return 0; +} + +static int __devinit snd_msnd_write_cfg_io1(int cfg, int num, u16 io) +{ + if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_IO1_BASEHI, HIBYTE(io))) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_IO1_BASELO, LOBYTE(io))) + return -EIO; + return 0; +} + +static int __devinit snd_msnd_write_cfg_irq(int cfg, int num, u16 irq) +{ + if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_IRQ_NUMBER, LOBYTE(irq))) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_IRQ_TYPE, IRQTYPE_EDGE)) + return -EIO; + return 0; +} + +static int __devinit snd_msnd_write_cfg_mem(int cfg, int num, int mem) +{ + u16 wmem; + + mem >>= 8; + wmem = (u16)(mem & 0xfff); + if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_MEMBASEHI, HIBYTE(wmem))) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_MEMBASELO, LOBYTE(wmem))) + return -EIO; + if (wmem && snd_msnd_write_cfg(cfg, IREG_MEMCONTROL, + MEMTYPE_HIADDR | MEMTYPE_16BIT)) + return -EIO; + return 0; +} + +static int __devinit snd_msnd_activate_logical(int cfg, int num) +{ + if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_ACTIVATE, LD_ACTIVATE)) + return -EIO; + return 0; +} + +static int __devinit snd_msnd_write_cfg_logical(int cfg, int num, u16 io0, + u16 io1, u16 irq, int mem) +{ + if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (snd_msnd_write_cfg_io0(cfg, num, io0)) + return -EIO; + if (snd_msnd_write_cfg_io1(cfg, num, io1)) + return -EIO; + if (snd_msnd_write_cfg_irq(cfg, num, irq)) + return -EIO; + if (snd_msnd_write_cfg_mem(cfg, num, mem)) + return -EIO; + if (snd_msnd_activate_logical(cfg, num)) + return -EIO; + return 0; +} + +static int __devinit snd_msnd_pinnacle_cfg_reset(int cfg) +{ + int i; + + /* Reset devices if told to */ + printk(KERN_INFO LOGNAME ": Resetting all devices\n"); + for (i = 0; i < 4; ++i) + if (snd_msnd_write_cfg_logical(cfg, i, 0, 0, 0, 0)) + return -EIO; + + return 0; +} +#endif + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ + +module_param_array(index, int, NULL, S_IRUGO); +MODULE_PARM_DESC(index, "Index value for msnd_pinnacle soundcard."); +module_param_array(id, charp, NULL, S_IRUGO); +MODULE_PARM_DESC(id, "ID string for msnd_pinnacle soundcard."); + +static long io[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT; +static int irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ; +static long mem[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT; + +static long cfg[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT; + +#ifndef MSND_CLASSIC +/* Extra Peripheral Configuration (Default: Disable) */ +static long ide_io0[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT; +static long ide_io1[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT; +static int ide_irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ; + +static long joystick_io[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT; +/* If we have the digital daugherboard... */ +static int digital[SNDRV_CARDS] __devinitdata; + +/* Extra Peripheral Configuration */ +static int reset[SNDRV_CARDS] __devinitdata; +#endif + +static int write_ndelay[SNDRV_CARDS] __devinitdata = + { [0 ... (SNDRV_CARDS-1)] = 1 }; + +#ifdef CONFIG_PNP +static int isapnp[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_ENABLE_PNP; +#endif + +#ifdef MODULE +MODULE_AUTHOR("Karsten Wiese <annabellesgarden(a)yahoo.de>"); +MODULE_DESCRIPTION("Turtle Beach " LONGNAME " Linux Driver"); +MODULE_LICENSE("GPL"); +MODULE_FIRMWARE(INITCODEFILE); +MODULE_FIRMWARE(PERMCODEFILE); + +static int calibrate_signal __devinitdata; + +module_param_array(io, long, NULL, S_IRUGO); +MODULE_PARM_DESC(io, "IO port #"); +module_param_array(irq, int, NULL, S_IRUGO); +module_param_array(mem, long, NULL, S_IRUGO); +module_param_array(write_ndelay, int, NULL, S_IRUGO); +module_param(calibrate_signal, int, S_IRUGO); +#ifndef MSND_CLASSIC +module_param_array(digital, int, NULL, S_IRUGO); +module_param_array(cfg, long, NULL, S_IRUGO); +module_param_array(reset, int, 0, S_IRUGO); +module_param_array(mpu_io, long, NULL, S_IRUGO); +module_param_array(mpu_irq, int, NULL, S_IRUGO); +module_param_array(ide_io0, long, NULL, S_IRUGO); +module_param_array(ide_io1, long, NULL, S_IRUGO); +module_param_array(ide_irq, int, NULL, S_IRUGO); +module_param_array(joystick_io, long, NULL, S_IRUGO); +#ifdef CONFIG_PNP +module_param_array(isapnp, bool, NULL, 0444); +MODULE_PARM_DESC(isapnp, "ISA PnP detection for specified soundcard."); +#endif +#endif + +#else /* not a module */ + +static int +calibrate_signal __devinitdata; +#endif /* MODULE */ + + +static int __devinit snd_msnd_isa_match(struct device *pdev, unsigned int i) +{ + if (io[i] == SNDRV_AUTO_PORT) + return 0; + + if (irq[i] == SNDRV_AUTO_PORT || mem[i] == SNDRV_AUTO_PORT) { + printk(KERN_WARNING LOGNAME ": io, irq and mem must be set\n"); + return 0; + } + +#ifdef MSND_CLASSIC + if (!(io[i] == 0x290 || + io[i] == 0x260 || + io[i] == 0x250 || + io[i] == 0x240 || + io[i] == 0x230 || + io[i] == 0x220 || + io[i] == 0x210 || + io[i] == 0x3e0)) { + printk(KERN_ERR LOGNAME ": \"io\" - DSP I/O base must be set " + " to 0x210, 0x220, 0x230, 0x240, 0x250, 0x260, 0x290, " + "or 0x3E0\n"); + return 0; + } +#else + if (io[i] < 0x100 || io[i] > 0x3e0 || (io[i] % 0x10) != 0) { + printk(KERN_ERR LOGNAME + ": \"io\" - DSP I/O base must within the range 0x100 " + "to 0x3E0 and must be evenly divisible by 0x10\n"); + return 0; + } +#endif /* MSND_CLASSIC */ + + if (!(irq[i] == 5 || + irq[i] == 7 || + irq[i] == 9 || + irq[i] == 10 || + irq[i] == 11 || + irq[i] == 12)) { + printk(KERN_ERR LOGNAME + ": \"irq\" - must be set to 5, 7, 9, 10, 11 or 12\n"); + return 0; + } + + if (!(mem[i] == 0xb0000 || + mem[i] == 0xc8000 || + mem[i] == 0xd0000 || + mem[i] == 0xd8000 || + mem[i] == 0xe0000 || + mem[i] == 0xe8000)) { + printk(KERN_ERR LOGNAME ": \"mem\" - must be set to " + "0xb0000, 0xc8000, 0xd0000, 0xd8000, 0xe0000 or " + "0xe8000\n"); + return 0; + } + +#ifndef MSND_CLASSIC + if (cfg[i] == SNDRV_AUTO_PORT) { + printk(KERN_INFO LOGNAME ": Assuming PnP mode\n"); + } else if (cfg[i] != 0x250 && cfg[i] != 0x260 && cfg[i] != 0x270) { + printk(KERN_INFO LOGNAME + ": Config port must be 0x250, 0x260 or 0x270 " + "(or unspecified for PnP mode)\n"); + return 0; + } +#endif /* MSND_CLASSIC */ + + return 1; +} + +static int __devinit snd_msnd_isa_probe(struct device *pdev, unsigned int idx) +{ + int err; + struct snd_card *card; + struct snd_msnd *chip; + + if (isapnp[idx] || cfg[idx] == SNDRV_AUTO_PORT) { + printk(KERN_INFO LOGNAME ": Assuming PnP mode\n"); + return -ENODEV; + } + + card = snd_card_new(index[idx], id[idx], THIS_MODULE, + sizeof(struct snd_msnd)); + if (!card) + return -ENOMEM; + + snd_card_set_dev(card, pdev); + chip = card->private_data; + chip->card = card; + +#ifdef MSND_CLASSIC + switch (irq[idx]) { + case 5: + chip->irqid = HPIRQ_5; break; + case 7: + chip->irqid = HPIRQ_7; break; + case 9: + chip->irqid = HPIRQ_9; break; + case 10: + chip->irqid = HPIRQ_10; break; + case 11: + chip->irqid = HPIRQ_11; break; + case 12: + chip->irqid = HPIRQ_12; break; + } + + switch (mem[idx]) { + case 0xb0000: + chip->memid = HPMEM_B000; break; + case 0xc8000: + chip->memid = HPMEM_C800; break; + case 0xd0000: + chip->memid = HPMEM_D000; break; + case 0xd8000: + chip->memid = HPMEM_D800; break; + case 0xe0000: + chip->memid = HPMEM_E000; break; + case 0xe8000: + chip->memid = HPMEM_E800; break; + } +#else + printk(KERN_INFO LOGNAME ": Non-PnP mode: configuring at port 0x%lx\n", + cfg[idx]); + + if (!request_region(cfg[idx], 2, "Pinnacle/Fiji Config")) { + printk(KERN_ERR LOGNAME ": Config port 0x%lx conflict\n", + cfg[idx]); + snd_card_free(card); + return -EIO; + } + if (reset[idx]) + if (snd_msnd_pinnacle_cfg_reset(cfg[idx])) { + err = -EIO; + goto cfg_error; + } + + /* DSP */ + err = snd_msnd_write_cfg_logical(cfg[idx], 0, + io[idx], 0, + irq[idx], mem[idx]); + + if (err) + goto cfg_error; + + /* The following are Pinnacle specific */ + + /* MPU */ + if (mpu_io[idx] != SNDRV_AUTO_PORT + && mpu_irq[idx] != SNDRV_AUTO_IRQ) { + printk(KERN_INFO LOGNAME + ": Configuring MPU to I/O 0x%lx IRQ %d\n", + mpu_io[idx], mpu_irq[idx]); + err = snd_msnd_write_cfg_logical(cfg[idx], 1, + mpu_io[idx], 0, + mpu_irq[idx], 0); + + if (err) + goto cfg_error; + } + + /* IDE */ + if (ide_io0[idx] != SNDRV_AUTO_PORT + && ide_io1[idx] != SNDRV_AUTO_PORT + && ide_irq[idx] != SNDRV_AUTO_IRQ) { + printk(KERN_INFO LOGNAME + ": Configuring IDE to I/O 0x%lx, 0x%lx IRQ %d\n", + ide_io0[idx], ide_io1[idx], ide_irq[idx]); + err = snd_msnd_write_cfg_logical(cfg[idx], 2, + ide_io0[idx], ide_io1[idx], + ide_irq[idx], 0); + + if (err) + goto cfg_error; + } + + /* Joystick */ + if (joystick_io[idx] != SNDRV_AUTO_PORT) { + printk(KERN_INFO LOGNAME + ": Configuring joystick to I/O 0x%lx\n", + joystick_io[idx]); + err = snd_msnd_write_cfg_logical(cfg[idx], 3, + joystick_io[idx], 0, + 0, 0); + + if (err) + goto cfg_error; + } + release_region(cfg[idx], 2); + +#endif /* MSND_CLASSIC */ + + set_default_audio_parameters(chip); +#ifdef MSND_CLASSIC + chip->type = msndClassic; +#else + chip->type = msndPinnacle; +#endif + chip->io = io[idx]; + chip->irq = irq[idx]; + chip->base = mem[idx]; + + chip->calibrate_signal = calibrate_signal ? 1 : 0; + chip->recsrc = 0; + chip->dspq_data_buff = DSPQ_DATA_BUFF; + chip->dspq_buff_size = DSPQ_BUFF_SIZE; + if (write_ndelay[idx]) + clear_bit(F_DISABLE_WRITE_NDELAY, &chip->flags); + else + set_bit(F_DISABLE_WRITE_NDELAY, &chip->flags); +#ifndef MSND_CLASSIC + if (digital[idx]) + set_bit(F_HAVEDIGITAL, &chip->flags); +#endif + spin_lock_init(&chip->lock); + err = snd_msnd_probe(card); + if (err < 0) { + printk(KERN_ERR LOGNAME ": Probe failed\n"); + snd_card_free(card); + return err; + } + + err = snd_msnd_attach(card); + if (err < 0) { + printk(KERN_ERR LOGNAME ": Attach failed\n"); + snd_card_free(card); + return err; + } + dev_set_drvdata(pdev, card); + + return 0; + +#ifndef MSND_CLASSIC +cfg_error: + release_region(cfg[idx], 2); + snd_card_free(card); + return err; +#endif +} + +static int __devexit snd_msnd_isa_remove(struct device *pdev, unsigned int dev) +{ + snd_msnd_unload(dev_get_drvdata(pdev)); + dev_set_drvdata(pdev, NULL); + return 0; +} + +#define DEV_NAME "msnd-pinnacle" + +static struct isa_driver snd_msnd_driver = { + .match = snd_msnd_isa_match, + .probe = snd_msnd_isa_probe, + .remove = __devexit_p(snd_msnd_isa_remove), + /* FIXME: suspend, resume */ + .driver = { + .name = DEV_NAME + }, +}; + +#ifdef CONFIG_PNP +static int __devinit snd_msnd_pnp_detect(struct pnp_card_link *pcard, + const struct pnp_card_device_id *pid) +{ + static int idx; + struct pnp_dev *pnp_dev; + struct pnp_dev *mpu_dev; + struct snd_card *card; + struct snd_msnd *chip; + int ret; + + for ( ; idx < SNDRV_CARDS; idx++) { + if (isapnp[idx]) + break; + } + if (idx >= SNDRV_CARDS) + return -ENODEV; + + /* + * Check that we still have room for another sound card ... + */ + pnp_dev = pnp_request_card_device(pcard, pid->devs[0].id, NULL); + if (!pnp_dev) + return -ENODEV; + + mpu_dev = pnp_request_card_device(pcard, pid->devs[1].id, NULL); + if (!mpu_dev) + return -ENODEV; + + if (!pnp_is_active(pnp_dev) && pnp_activate_dev(pnp_dev) < 0) { + printk(KERN_INFO "msnd_pinnacle: device is inactive\n"); + return -EBUSY; + } + + if (!pnp_is_active(mpu_dev) && pnp_activate_dev(mpu_dev) < 0) { + printk(KERN_INFO "msnd_pinnacle: MPU device is inactive\n"); + return -EBUSY; + } + + /* + * Create a new ALSA sound card entry, in anticipation + * of detecting our hardware ... + */ + card = snd_card_new(index[idx], id[idx], THIS_MODULE, + sizeof(struct snd_msnd)); + if (!card) + return -ENOMEM; + + chip = card->private_data; + chip->card = card; + snd_card_set_dev(card, &pcard->card->dev); + + /* + * Read the correct parameters off the ISA PnP bus ... + */ + io[idx] = pnp_port_start(pnp_dev, 0); + irq[idx] = pnp_irq(pnp_dev, 0); + mem[idx] = pnp_mem_start(pnp_dev, 0); + mpu_io[idx] = pnp_port_start(mpu_dev, 0); + mpu_irq[idx] = pnp_irq(mpu_dev, 0); + + set_default_audio_parameters(chip); +#ifdef MSND_CLASSIC + chip->type = msndClassic; +#else + chip->type = msndPinnacle; +#endif + chip->io = io[idx]; + chip->irq = irq[idx]; + chip->base = mem[idx]; + + chip->calibrate_signal = calibrate_signal ? 1 : 0; + chip->recsrc = 0; + chip->dspq_data_buff = DSPQ_DATA_BUFF; + chip->dspq_buff_size = DSPQ_BUFF_SIZE; + if (write_ndelay[idx]) + clear_bit(F_DISABLE_WRITE_NDELAY, &chip->flags); + else + set_bit(F_DISABLE_WRITE_NDELAY, &chip->flags); +#ifndef MSND_CLASSIC + if (digital[idx]) + set_bit(F_HAVEDIGITAL, &chip->flags); +#endif + spin_lock_init(&chip->lock); + ret = snd_msnd_probe(card); + if (ret < 0) { + printk(KERN_ERR LOGNAME ": Probe failed\n"); + goto _release_card; + } + + ret = snd_msnd_attach(card); + if (ret < 0) { + printk(KERN_ERR LOGNAME ": Attach failed\n"); + goto _release_card; + } + + pnp_set_card_drvdata(pcard, card); + ++idx; + return 0; + +_release_card: + snd_card_free(card); + return ret; +} + +static void __devexit snd_msnd_pnp_remove(struct pnp_card_link *pcard) +{ + snd_msnd_unload(pnp_get_card_drvdata(pcard)); + pnp_set_card_drvdata(pcard, NULL); +} + +static int isa_registered; +static int pnp_registered; + +static struct pnp_card_device_id msnd_pnpids[] = { + /* Pinnacle PnP */ + { .id = "BVJ0440", .devs = { { "TBS0000" }, { "TBS0001" } } }, + { .id = "" } /* end */ +}; + +MODULE_DEVICE_TABLE(pnp_card, msnd_pnpids); + +static struct pnp_card_driver msnd_pnpc_driver = { + .flags = PNP_DRIVER_RES_DO_NOT_CHANGE, + .name = "msnd_pinnacle", + .id_table = msnd_pnpids, + .probe = snd_msnd_pnp_detect, + .remove = __devexit_p(snd_msnd_pnp_remove), +}; +#endif /* CONFIG_PNP */ + +static int __init snd_msnd_init(void) +{ + int err; + + err = isa_register_driver(&snd_msnd_driver, SNDRV_CARDS); +#ifdef CONFIG_PNP + if (!err) + isa_registered = 1; + + err = pnp_register_card_driver(&msnd_pnpc_driver); + if (!err) + pnp_registered = 1; + + if (isa_registered) + err = 0; +#endif + return err; +} + +static void __exit snd_msnd_exit(void) +{ +#ifdef CONFIG_PNP + if (pnp_registered) + pnp_unregister_card_driver(&msnd_pnpc_driver); + if (isa_registered) +#endif + isa_unregister_driver(&snd_msnd_driver); +} + +module_init(snd_msnd_init); +module_exit(snd_msnd_exit); + diff -urpN linux-rc5/sound/isa/msnd/msnd_pinnacle.h linux-git/sound/isa/msnd/msnd_pinnacle.h --- linux-rc5/sound/isa/msnd/msnd_pinnacle.h 1970-01-01 01:00:00.000000000 +0100 +++ linux-git/sound/isa/msnd/msnd_pinnacle.h 2009-01-22 21:03:22.885084729 +0100 @@ -0,0 +1,203 @@ +/********************************************************************* + * + * msnd_pinnacle.h + * + * Turtle Beach MultiSound Sound Card Driver for Linux + * + * Some parts of this header file were derived from the Turtle Beach + * MultiSound Driver Development Kit. + * + * Copyright (C) 1998 Andrew Veliath + * Copyright (C) 1993 Turtle Beach Systems, Inc. + * + * 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., 675 Mass Ave, Cambridge, MA 02139, USA. + * + ********************************************************************/ +#ifndef __MSND_PINNACLE_H +#define __MSND_PINNACLE_H + +#define DSP_NUMIO 0x08 + +#define IREG_LOGDEVICE 0x07 +#define IREG_ACTIVATE 0x30 +#define LD_ACTIVATE 0x01 +#define LD_DISACTIVATE 0x00 +#define IREG_EECONTROL 0x3F +#define IREG_MEMBASEHI 0x40 +#define IREG_MEMBASELO 0x41 +#define IREG_MEMCONTROL 0x42 +#define IREG_MEMRANGEHI 0x43 +#define IREG_MEMRANGELO 0x44 +#define MEMTYPE_8BIT 0x00 +#define MEMTYPE_16BIT 0x02 +#define MEMTYPE_RANGE 0x00 +#define MEMTYPE_HIADDR 0x01 +#define IREG_IO0_BASEHI 0x60 +#define IREG_IO0_BASELO 0x61 +#define IREG_IO1_BASEHI 0x62 +#define IREG_IO1_BASELO 0x63 +#define IREG_IRQ_NUMBER 0x70 +#define IREG_IRQ_TYPE 0x71 +#define IRQTYPE_HIGH 0x02 +#define IRQTYPE_LOW 0x00 +#define IRQTYPE_LEVEL 0x01 +#define IRQTYPE_EDGE 0x00 + +#define HP_DSPR 0x04 +#define HP_BLKS 0x04 + +#define HPDSPRESET_OFF 2 +#define HPDSPRESET_ON 0 + +#define HPBLKSEL_0 2 +#define HPBLKSEL_1 3 + +#define HIMT_DAT_OFF 0x03 + +#define HIDSP_PLAY_UNDER 0x00 +#define HIDSP_INT_PLAY_UNDER 0x01 +#define HIDSP_SSI_TX_UNDER 0x02 +#define HIDSP_RECQ_OVERFLOW 0x08 +#define HIDSP_INT_RECORD_OVER 0x09 +#define HIDSP_SSI_RX_OVERFLOW 0x0a + +#define HIDSP_MIDI_IN_OVER 0x10 + +#define HIDSP_MIDI_FRAME_ERR 0x11 +#define HIDSP_MIDI_PARITY_ERR 0x12 +#define HIDSP_MIDI_OVERRUN_ERR 0x13 + +#define HIDSP_INPUT_CLIPPING 0x20 +#define HIDSP_MIX_CLIPPING 0x30 +#define HIDSP_DAT_IN_OFF 0x21 + +#define HDEXAR_SET_ANA_IN 0 +#define HDEXAR_CLEAR_PEAKS 1 +#define HDEXAR_IN_SET_POTS 2 +#define HDEXAR_AUX_SET_POTS 3 +#define HDEXAR_CAL_A_TO_D 4 +#define HDEXAR_RD_EXT_DSP_BITS 5 + +#define HDEXAR_SET_SYNTH_IN 4 +#define HDEXAR_READ_DAT_IN 5 +#define HDEXAR_MIC_SET_POTS 6 +#define HDEXAR_SET_DAT_IN 7 + +#define HDEXAR_SET_SYNTH_48 8 +#define HDEXAR_SET_SYNTH_44 9 + +#define TIME_PRO_RESET_DONE 0x028A +#define TIME_PRO_SYSEX 0x001E +#define TIME_PRO_RESET 0x0032 + +#define DAR_BUFF_SIZE 0x1000 + +#define MIDQ_BUFF_SIZE 0x800 +#define DSPQ_BUFF_SIZE 0x5A0 + +#define DSPQ_DATA_BUFF 0x7860 + +#define MOP_WAVEHDR 0 +#define MOP_EXTOUT 1 +#define MOP_HWINIT 0xfe +#define MOP_NONE 0xff +#define MOP_MAX 1 + +#define MIP_EXTIN 0 +#define MIP_WAVEHDR 1 +#define MIP_HWINIT 0xfe +#define MIP_MAX 1 + +/* Pinnacle/Fiji SMA Common Data */ +#define SMA_wCurrPlayBytes 0x0000 +#define SMA_wCurrRecordBytes 0x0002 +#define SMA_wCurrPlayVolLeft 0x0004 +#define SMA_wCurrPlayVolRight 0x0006 +#define SMA_wCurrInVolLeft 0x0008 +#define SMA_wCurrInVolRight 0x000a +#define SMA_wCurrMHdrVolLeft 0x000c +#define SMA_wCurrMHdrVolRight 0x000e +#define SMA_dwCurrPlayPitch 0x0010 +#define SMA_dwCurrPlayRate 0x0014 +#define SMA_wCurrMIDIIOPatch 0x0018 +#define SMA_wCurrPlayFormat 0x001a +#define SMA_wCurrPlaySampleSize 0x001c +#define SMA_wCurrPlayChannels 0x001e +#define SMA_wCurrPlaySampleRate 0x0020 +#define SMA_wCurrRecordFormat 0x0022 +#define SMA_wCurrRecordSampleSize 0x0024 +#define SMA_wCurrRecordChannels 0x0026 +#define SMA_wCurrRecordSampleRate 0x0028 +#define SMA_wCurrDSPStatusFlags 0x002a +#define SMA_wCurrHostStatusFlags 0x002c +#define SMA_wCurrInputTagBits 0x002e +#define SMA_wCurrLeftPeak 0x0030 +#define SMA_wCurrRightPeak 0x0032 +#define SMA_bMicPotPosLeft 0x0034 +#define SMA_bMicPotPosRight 0x0035 +#define SMA_bMicPotMaxLeft 0x0036 +#define SMA_bMicPotMaxRight 0x0037 +#define SMA_bInPotPosLeft 0x0038 +#define SMA_bInPotPosRight 0x0039 +#define SMA_bAuxPotPosLeft 0x003a +#define SMA_bAuxPotPosRight 0x003b +#define SMA_bInPotMaxLeft 0x003c +#define SMA_bInPotMaxRight 0x003d +#define SMA_bAuxPotMaxLeft 0x003e +#define SMA_bAuxPotMaxRight 0x003f +#define SMA_bInPotMaxMethod 0x0040 +#define SMA_bAuxPotMaxMethod 0x0041 +#define SMA_wCurrMastVolLeft 0x0042 +#define SMA_wCurrMastVolRight 0x0044 +#define SMA_wCalFreqAtoD 0x0046 +#define SMA_wCurrAuxVolLeft 0x0048 +#define SMA_wCurrAuxVolRight 0x004a +#define SMA_wCurrPlay1VolLeft 0x004c +#define SMA_wCurrPlay1VolRight 0x004e +#define SMA_wCurrPlay2VolLeft 0x0050 +#define SMA_wCurrPlay2VolRight 0x0052 +#define SMA_wCurrPlay3VolLeft 0x0054 +#define SMA_wCurrPlay3VolRight 0x0056 +#define SMA_wCurrPlay4VolLeft 0x0058 +#define SMA_wCurrPlay4VolRight 0x005a +#define SMA_wCurrPlay1PeakLeft 0x005c +#define SMA_wCurrPlay1PeakRight 0x005e +#define SMA_wCurrPlay2PeakLeft 0x0060 +#define SMA_wCurrPlay2PeakRight 0x0062 +#define SMA_wCurrPlay3PeakLeft 0x0064 +#define SMA_wCurrPlay3PeakRight 0x0066 +#define SMA_wCurrPlay4PeakLeft 0x0068 +#define SMA_wCurrPlay4PeakRight 0x006a +#define SMA_wCurrPlayPeakLeft 0x006c +#define SMA_wCurrPlayPeakRight 0x006e +#define SMA_wCurrDATSR 0x0070 +#define SMA_wCurrDATRXCHNL 0x0072 +#define SMA_wCurrDATTXCHNL 0x0074 +#define SMA_wCurrDATRXRate 0x0076 +#define SMA_dwDSPPlayCount 0x0078 +#define SMA__size 0x007c + +#define INITCODEFILE "turtlebeach/pndspini.bin" +#define PERMCODEFILE "turtlebeach/pndsperm.bin" +#define LONGNAME "MultiSound (Pinnacle/Fiji)" + +int snd_msnd_send_dsp_cmd_chk(struct snd_msnd *chip, register u8 cmd); + +/* functions from external mixer file */ +void snd_msndmix_setup(struct snd_msnd *chip); +unsigned long snd_msndmix_force_recsrc(struct snd_msnd *chip, int recsrc); +int __devinit snd_msndmix_new(struct snd_card *card); + +#endif /* __MSND_PINNACLE_H */ diff -urpN linux-rc5/sound/isa/msnd/msnd_pinnacle_mixer.c linux-git/sound/isa/msnd/msnd_pinnacle_mixer.c --- linux-rc5/sound/isa/msnd/msnd_pinnacle_mixer.c 1970-01-01 01:00:00.000000000 +0100 +++ linux-git/sound/isa/msnd/msnd_pinnacle_mixer.c 2009-01-22 21:01:42.220129702 +0100 @@ -0,0 +1,345 @@ +/*************************************************************************** + msnd_pinnacle_mixer.c - description + ------------------- + begin : Fre Jun 7 2002 + copyright : (C) 2002 by karsten wiese + email : annabellesgarden(a)yahoo.de + ***************************************************************************/ + +/*************************************************************************** + * * + * This program is free software; you can redistribute it and/or modify * + * it under the terms of the GNU General Public License as published by * + * the Free Software Foundation; either version 2 of the License, or * + * (at your option) any later version. * + * * + ***************************************************************************/ + +#include <linux/io.h> + +#include <sound/core.h> +#include <sound/control.h> +#include "msnd.h" +#include "msnd_pinnacle.h" + + +#define MSND_MIXER_VOLUME 0 +#define MSND_MIXER_PCM 1 +#define MSND_MIXER_AUX 2 /* Input source 1 (aux1) */ +#define MSND_MIXER_IMIX 3 /* Recording monitor */ +#define MSND_MIXER_SYNTH 4 +#define MSND_MIXER_SPEAKER 5 +#define MSND_MIXER_LINE 6 +#define MSND_MIXER_MIC 7 +#define MSND_MIXER_RECLEV 11 /* Recording level */ +#define MSND_MIXER_IGAIN 12 /* Input gain */ +#define MSND_MIXER_OGAIN 13 /* Output gain */ +#define MSND_MIXER_DIGITAL 17 /* Digital (input) 1 */ + +/* Device mask bits */ + +#define MSND_MASK_VOLUME (1 << MSND_MIXER_VOLUME) +#define MSND_MASK_SYNTH (1 << MSND_MIXER_SYNTH) +#define MSND_MASK_PCM (1 << MSND_MIXER_PCM) +#define MSND_MASK_SPEAKER (1 << MSND_MIXER_SPEAKER) +#define MSND_MASK_LINE (1 << MSND_MIXER_LINE) +#define MSND_MASK_MIC (1 << MSND_MIXER_MIC) +#define MSND_MASK_IMIX (1 << MSND_MIXER_IMIX) +#define MSND_MASK_RECLEV (1 << MSND_MIXER_RECLEV) +#define MSND_MASK_IGAIN (1 << MSND_MIXER_IGAIN) +#define MSND_MASK_OGAIN (1 << MSND_MIXER_OGAIN) +#define MSND_MASK_AUX (1 << MSND_MIXER_AUX) +#define MSND_MASK_DIGITAL (1 << MSND_MIXER_DIGITAL) + +static int snd_msndmix_info_mux(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[3] = { + "Analog", "MASS", "SPDIF", + }; + struct snd_msnd *chip = snd_kcontrol_chip(kcontrol); + unsigned items = test_bit(F_HAVEDIGITAL, &chip->flags) ? 3 : 2; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = items; + if (uinfo->value.enumerated.item >= items) + uinfo->value.enumerated.item = items - 1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + return 0; +} + +static int snd_msndmix_get_mux(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_msnd *chip = snd_kcontrol_chip(kcontrol); + /* MSND_MASK_IMIX is the default */ + ucontrol->value.enumerated.item[0] = 0; + + if (chip->recsrc & MSND_MASK_SYNTH) { + ucontrol->value.enumerated.item[0] = 1; + } else if ((chip->recsrc & MSND_MASK_DIGITAL) && + test_bit(F_HAVEDIGITAL, &chip->flags)) { + ucontrol->value.enumerated.item[0] = 2; + } + + + return 0; +} + +static int snd_msndmix_set_mux(struct snd_msnd *chip, int val) +{ + unsigned newrecsrc; + int change; + unsigned char msndbyte; + + switch (val) { + case 0: + newrecsrc = MSND_MASK_IMIX; + msndbyte = HDEXAR_SET_ANA_IN; + break; + case 1: + newrecsrc = MSND_MASK_SYNTH; + msndbyte = HDEXAR_SET_SYNTH_IN; + break; + case 2: + newrecsrc = MSND_MASK_DIGITAL; + msndbyte = HDEXAR_SET_DAT_IN; + break; + default: + return -EINVAL; + } + change = newrecsrc != chip->recsrc; + if (change) { + change = 0; + if (!snd_msnd_send_word(chip, 0, 0, msndbyte)) + if (!snd_msnd_send_dsp_cmd_chk(chip, HDEX_AUX_REQ)) { + chip->recsrc = newrecsrc; + change = 1; + } + } + return change; +} + + +static int snd_msndmix_put_mux(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_msnd *msnd = snd_kcontrol_chip(kcontrol); + return snd_msndmix_set_mux(msnd, ucontrol->value.enumerated.item[0]); +} + + +static int snd_msndmix_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 100; + return 0; +} + +static int snd_msndmix_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_msnd *msnd = snd_kcontrol_chip(kcontrol); + int addr = kcontrol->private_value; + unsigned long flags; + + spin_lock_irqsave(&msnd->mixer_lock, flags); + ucontrol->value.integer.value[0] = msnd->left_levels[addr] * 100; + ucontrol->value.integer.value[0] /= 0xFFFF; + ucontrol->value.integer.value[1] = msnd->right_levels[addr] * 100; + ucontrol->value.integer.value[1] /= 0xFFFF; + spin_unlock_irqrestore(&msnd->mixer_lock, flags); + return 0; +} + + +#define update_volm(a, b) \ + do { \ + writew((dev->left_levels[a] >> 1) * \ + readw(dev->SMA + SMA_wCurrMastVolLeft) / 0xffff, \ + dev->SMA + SMA_##b##Left); \ + writew((dev->right_levels[a] >> 1) * \ + readw(dev->SMA + SMA_wCurrMastVolRight) / 0xffff, \ + dev->SMA + SMA_##b##Right); \ + } while (0); + +#define update_potm(d, s, ar) \ + do { \ + writeb((dev->left_levels[d] >> 8) * \ + readw(dev->SMA + SMA_wCurrMastVolLeft) / 0xffff, \ + dev->SMA + SMA_##s##Left); \ + writeb((dev->right_levels[d] >> 8) * \ + readw(dev->SMA + SMA_wCurrMastVolRight) / 0xffff, \ + dev->SMA + SMA_##s##Right); \ + if (snd_msnd_send_word(dev, 0, 0, ar) == 0) \ + snd_msnd_send_dsp_cmd_chk(dev, HDEX_AUX_REQ); \ + } while (0); + +#define update_pot(d, s, ar) \ + do { \ + writeb(dev->left_levels[d] >> 8, \ + dev->SMA + SMA_##s##Left); \ + writeb(dev->right_levels[d] >> 8, \ + dev->SMA + SMA_##s##Right); \ + if (snd_msnd_send_word(dev, 0, 0, ar) == 0) \ + snd_msnd_send_dsp_cmd_chk(dev, HDEX_AUX_REQ); \ + } while (0); + + +static int snd_msndmix_set(struct snd_msnd *dev, int d, int left, int right) +{ + int bLeft, bRight; + int wLeft, wRight; + int updatemaster = 0; + + if (d >= LEVEL_ENTRIES) + return -EINVAL; + + bLeft = left * 0xff / 100; + wLeft = left * 0xffff / 100; + + bRight = right * 0xff / 100; + wRight = right * 0xffff / 100; + + dev->left_levels[d] = wLeft; + dev->right_levels[d] = wRight; + + switch (d) { + /* master volume unscaled controls */ + case MSND_MIXER_LINE: /* line pot control */ + /* scaled by IMIX in digital mix */ + writeb(bLeft, dev->SMA + SMA_bInPotPosLeft); + writeb(bRight, dev->SMA + SMA_bInPotPosRight); + if (snd_msnd_send_word(dev, 0, 0, HDEXAR_IN_SET_POTS) == 0) + snd_msnd_send_dsp_cmd_chk(dev, HDEX_AUX_REQ); + break; +#ifndef MSND_CLASSIC + case MSND_MIXER_MIC: /* mic pot control */ + /* scaled by IMIX in digital mix */ + writeb(bLeft, dev->SMA + SMA_bMicPotPosLeft); + writeb(bRight, dev->SMA + SMA_bMicPotPosRight); + if (snd_msnd_send_word(dev, 0, 0, HDEXAR_MIC_SET_POTS) == 0) + snd_msnd_send_dsp_cmd_chk(dev, HDEX_AUX_REQ); + break; +#endif + case MSND_MIXER_VOLUME: /* master volume */ + writew(wLeft, dev->SMA + SMA_wCurrMastVolLeft); + writew(wRight, dev->SMA + SMA_wCurrMastVolRight); + /* fall through */ + + case MSND_MIXER_AUX: /* aux pot control */ + /* scaled by master volume */ + /* fall through */ + + /* digital controls */ + case MSND_MIXER_SYNTH: /* synth vol (dsp mix) */ + case MSND_MIXER_PCM: /* pcm vol (dsp mix) */ + case MSND_MIXER_IMIX: /* input monitor (dsp mix) */ + /* scaled by master volume */ + updatemaster = 1; + break; + + default: + return 0; + } + + if (updatemaster) { + /* update master volume scaled controls */ + update_volm(MSND_MIXER_PCM, wCurrPlayVol); + update_volm(MSND_MIXER_IMIX, wCurrInVol); +#ifndef MSND_CLASSIC + update_volm(MSND_MIXER_SYNTH, wCurrMHdrVol); +#endif + update_potm(MSND_MIXER_AUX, bAuxPotPos, HDEXAR_AUX_SET_POTS); + } + + return 0; +} + +static int snd_msndmix_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_msnd *msnd = snd_kcontrol_chip(kcontrol); + int change, addr = kcontrol->private_value; + int left, right; + unsigned long flags; + + left = ucontrol->value.integer.value[0] % 101; + right = ucontrol->value.integer.value[1] % 101; + spin_lock_irqsave(&msnd->mixer_lock, flags); + change = msnd->left_levels[addr] != left + || msnd->right_levels[addr] != right; + snd_msndmix_set(msnd, addr, left, right); + spin_unlock_irqrestore(&msnd->mixer_lock, flags); + return change; +} + + +#define DUMMY_VOLUME(xname, xindex, addr) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .info = snd_msndmix_volume_info, \ + .get = snd_msndmix_volume_get, .put = snd_msndmix_volume_put, \ + .private_value = addr } + + +static struct snd_kcontrol_new snd_msnd_controls[] = { +DUMMY_VOLUME("Master Volume", 0, MSND_MIXER_VOLUME), +DUMMY_VOLUME("PCM Volume", 0, MSND_MIXER_PCM), +DUMMY_VOLUME("Aux Volume", 0, MSND_MIXER_AUX), +DUMMY_VOLUME("Line Volume", 0, MSND_MIXER_LINE), +DUMMY_VOLUME("Mic Volume", 0, MSND_MIXER_MIC), +DUMMY_VOLUME("Monitor", 0, MSND_MIXER_IMIX), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = snd_msndmix_info_mux, + .get = snd_msndmix_get_mux, + .put = snd_msndmix_put_mux, +} +}; + + +int __devinit snd_msndmix_new(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + unsigned int idx; + int err; + + if (snd_BUG_ON(!chip)) + return -EINVAL; + spin_lock_init(&chip->mixer_lock); + strcpy(card->mixername, "MSND Pinnacle Mixer"); + + for (idx = 0; idx < ARRAY_SIZE(snd_msnd_controls); idx++) + err = snd_ctl_add(card, + snd_ctl_new1(snd_msnd_controls + idx, chip)); + if (err < 0) + return err; + + return 0; +} + +void snd_msndmix_setup(struct snd_msnd *dev) +{ + update_pot(MSND_MIXER_LINE, bInPotPos, HDEXAR_IN_SET_POTS); + update_potm(MSND_MIXER_AUX, bAuxPotPos, HDEXAR_AUX_SET_POTS); + update_volm(MSND_MIXER_PCM, wCurrPlayVol); + update_volm(MSND_MIXER_IMIX, wCurrInVol); +#ifndef MSND_CLASSIC + update_pot(MSND_MIXER_MIC, bMicPotPos, HDEXAR_MIC_SET_POTS); + update_volm(MSND_MIXER_SYNTH, wCurrMHdrVol); +#endif + +} + +unsigned long snd_msndmix_force_recsrc(struct snd_msnd *dev, int recsrc) +{ + dev->recsrc = -1; + return snd_msndmix_set_mux(dev, recsrc); +} + 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2 3

[alsa-devel] [PATCH v2 2.6.30] ASoC: remove stand-alone mode support from CS4270 codec driver
by Timur Tabi 23 Jan '09

23 Jan '09

The CS4270 supports stand-alone mode, where the codec is not connect to the I2C or SPI buses. Instead, input voltages configure the codec at power-on. The CS4270 ASoC device driver has partial support for this mode, but the code was never tested, and partial support doesn't help anyone. It also made the rest of the code more complicated than necessary. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- v2: Added Kconfig changes sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/cs4270.c | 92 +++++++++++++------------------------------- sound/soc/fsl/Kconfig | 3 +- 3 files changed, 30 insertions(+), 66 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cb5fcd6..136e1bc 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -65,6 +65,7 @@ config SND_SOC_AK4535 # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate + depends on I2C # Cirrus Logic CS4270 Codec Hardware Mute Support # Select if you have external muting circuitry attached to your CS4270. diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index f1aa0c3..2e4ce04 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -12,11 +12,7 @@ * * Current features/limitations: * - * 1) Software mode is supported. Stand-alone mode is automatically - * selected if I2C is disabled or if a CS4270 is not found on the I2C - * bus. However, stand-alone mode is only partially implemented because - * there is no mechanism yet for this driver and the machine driver to - * communicate the values of the M0, M1, MCLK1, and MCLK2 pins. + * 1) Software mode is supported. Stand-alone mode is not supported. * 2) Only I2C is supported, not SPI * 3) Only Master mode is supported, not Slave. * 4) The machine driver's 'startup' function must call @@ -33,14 +29,6 @@ #include <sound/initval.h> #include <linux/i2c.h> -#include "cs4270.h" - -/* If I2C is defined, then we support software mode. However, if we're - not compiled as module but I2C is, then we can't use I2C calls. */ -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -#define USE_I2C -#endif - /* Private data for the CS4270 */ struct cs4270_private { unsigned int mclk; /* Input frequency of the MCLK pin */ @@ -60,8 +48,6 @@ struct cs4270_private { SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE) -#ifdef USE_I2C - /* CS4270 registers addresses */ #define CS4270_CHIPID 0x01 /* Chip ID */ #define CS4270_PWRCTL 0x02 /* Power Control */ @@ -272,17 +258,6 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, } /* - * A list of addresses on which this CS4270 could use. I2C addresses are - * 7 bits. For the CS4270, the upper four bits are always 1001, and the - * lower three bits are determined via the AD2, AD1, and AD0 pins - * (respectively). - */ -static const unsigned short normal_i2c[] = { - 0x48, 0x49, 0x4A, 0x4B, 0x4C, 0x4D, 0x4E, 0x4F, I2C_CLIENT_END -}; -I2C_CLIENT_INSMOD; - -/* * Pre-fill the CS4270 register cache. * * We use the auto-increment feature of the CS4270 to read all registers in @@ -476,7 +451,6 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, } #ifdef CONFIG_SND_SOC_CS4270_HWMUTE - /* * Set the CS4270 external mute * @@ -501,32 +475,16 @@ static int cs4270_mute(struct snd_soc_dai *dai, int mute) return snd_soc_write(codec, CS4270_MUTE, reg6); } - +#else +#define cs4270_mute NULL #endif -static int cs4270_i2c_probe(struct i2c_client *, const struct i2c_device_id *); - /* A list of non-DAPM controls that the CS4270 supports */ static const struct snd_kcontrol_new cs4270_snd_controls[] = { SOC_DOUBLE_R("Master Playback Volume", CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 1) }; -static const struct i2c_device_id cs4270_id[] = { - {"cs4270", 0}, - {} -}; -MODULE_DEVICE_TABLE(i2c, cs4270_id); - -static struct i2c_driver cs4270_i2c_driver = { - .driver = { - .name = "CS4270 I2C", - .owner = THIS_MODULE, - }, - .id_table = cs4270_id, - .probe = cs4270_i2c_probe, -}; - /* * Global variable to store socdev for i2c probe function. * @@ -633,7 +591,20 @@ error: return ret; } -#endif /* USE_I2C*/ +static const struct i2c_device_id cs4270_id[] = { + {"cs4270", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs4270_id); + +static struct i2c_driver cs4270_i2c_driver = { + .driver = { + .name = "cs4270", + .owner = THIS_MODULE, + }, + .id_table = cs4270_id, + .probe = cs4270_i2c_probe, +}; struct snd_soc_dai cs4270_dai = { .name = "CS4270", @@ -698,7 +669,6 @@ static int cs4270_probe(struct platform_device *pdev) goto error_free_codec; } -#ifdef USE_I2C cs4270_socdev = socdev; ret = i2c_add_driver(&cs4270_i2c_driver); @@ -708,20 +678,16 @@ static int cs4270_probe(struct platform_device *pdev) } /* Did we find a CS4270 on the I2C bus? */ - if (codec->control_data) { - /* Initialize codec ops */ - cs4270_dai.ops.hw_params = cs4270_hw_params; - cs4270_dai.ops.set_sysclk = cs4270_set_dai_sysclk; - cs4270_dai.ops.set_fmt = cs4270_set_dai_fmt; -#ifdef CONFIG_SND_SOC_CS4270_HWMUTE - cs4270_dai.ops.digital_mute = cs4270_mute; -#endif - } else - printk(KERN_INFO "cs4270: no I2C device found, " - "using stand-alone mode\n"); -#else - printk(KERN_INFO "cs4270: I2C disabled, using stand-alone mode\n"); -#endif + if (!codec->control_data) { + printk(KERN_ERR "cs4270: failed to attach driver"); + goto error_del_driver; + } + + /* Initialize codec ops */ + cs4270_dai.ops.hw_params = cs4270_hw_params; + cs4270_dai.ops.set_sysclk = cs4270_set_dai_sysclk; + cs4270_dai.ops.set_fmt = cs4270_set_dai_fmt; + cs4270_dai.ops.digital_mute = cs4270_mute; ret = snd_soc_init_card(socdev); if (ret < 0) { @@ -732,11 +698,9 @@ static int cs4270_probe(struct platform_device *pdev) return 0; error_del_driver: -#ifdef USE_I2C i2c_del_driver(&cs4270_i2c_driver); error_free_pcms: -#endif snd_soc_free_pcms(socdev); error_free_codec: @@ -752,9 +716,7 @@ static int cs4270_remove(struct platform_device *pdev) snd_soc_free_pcms(socdev); -#ifdef USE_I2C i2c_del_driver(&cs4270_i2c_driver); -#endif kfree(socdev->codec); socdev->codec = NULL; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index c7c78c3..9fc9082 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -10,7 +10,8 @@ config SND_SOC_MPC8610 config SND_SOC_MPC8610_HPCD tristate "ALSA SoC support for the Freescale MPC8610 HPCD board" - depends on MPC8610_HPCD + # I2C is necessary for the CS4270 driver + depends on MPC8610_HPCD && I2C select SND_SOC_MPC8610 select SND_SOC_CS4270 select SND_SOC_CS4270_VD33_ERRATA -- 1.5.5

3 5

[alsa-devel] ALSA device list: No soundcards found.
by Justin P. Mattock 23 Jan '09

23 Jan '09

when using any kernel(2.6.17-present) I see under dmesg: [ 2.702799] Advanced Linux Sound Architecture Driver Version 1.0.18a. [ 2.711657] ALSA device list: [ 2.720204] No soundcards found. how do I change this? i.g. ALSA device list: (some card name) I have a macbook pro ATI chipset. regards; Justin P. Mattock

2 1

[alsa-devel] [PATCH] ASoC: Add missing comma to SND_SOC_DAPM_SWITCH_E in soc-dapm.h
by Peter Ujfalusi 23 Jan '09

23 Jan '09

Typo fix. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- include/sound/soc-dapm.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 0accdba..bb3a863 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -118,7 +118,7 @@ #define SND_SOC_DAPM_SWITCH_E(wname, wreg, wshift, winvert, wcontrols, \ wevent, wflags) \ { .id = snd_soc_dapm_switch, .name = wname, .reg = wreg, .shift = wshift, \ - .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = 1 \ + .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = 1, \ .event = wevent, .event_flags = wflags} #define SND_SOC_DAPM_MUX_E(wname, wreg, wshift, winvert, wcontrols, \ wevent, wflags) \ -- 1.6.1

2 1

[alsa-devel] [PATCH] hda: add reference board SND_PCI_QUIRK
by Matthew Ranostay 23 Jan '09

23 Jan '09

Add another LanParty reference board SND_PCI_QUIRK to quirk lists of all codec families. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index dbe8b12..032dd0b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1510,6 +1510,8 @@ static struct snd_pci_quirk stac9200_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_REF), /* Dell laptops have BIOS problem */ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01a8, "unknown Dell", STAC_9200_DELL_D21), @@ -1659,6 +1661,7 @@ static struct snd_pci_quirk stac925x_codec_id_cfg_tbl[] = { static struct snd_pci_quirk stac925x_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_REF), SND_PCI_QUIRK(0x8384, 0x7632, "Stac9202 Reference Board", STAC_REF), /* Default table for unknown ID */ @@ -1702,6 +1705,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_92HD73XX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_92HD73XX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0254, "Dell Studio 1535", STAC_DELL_M6_DMIC), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0255, @@ -1744,6 +1749,8 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_92HD83XXX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_92HD83XXX_REF), {} /* terminator */ }; @@ -1793,6 +1800,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_92HD71BXX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_92HD71BXX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f2, "HP dv5", STAC_HP_M4), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f4, @@ -1981,6 +1990,8 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_D945_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_D945_REF), /* Intel 945G based systems */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0101, "Intel D945G", STAC_D945GTP3), @@ -2137,6 +2148,8 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_D965_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_D965_REF), /* Intel 946 based systems */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x3d01, "Intel D946", STAC_D965_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0xa301, "Intel D946", STAC_D965_3ST), @@ -2246,6 +2259,8 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_9205_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_9205_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f1, "unknown Dell", STAC_9205_DELL_M42), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f2,

2 1

[alsa-devel] [PATCH] atiixp: change parameter to macro define
by Matthew Ranostay 23 Jan '09

23 Jan '09

Change 0x15bd parameter to PCI_VENDOR_ID_DFI define. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 9ce8548..6baa6c4 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -297,7 +297,7 @@ static struct pci_device_id snd_atiixp_ids[] = { MODULE_DEVICE_TABLE(pci, snd_atiixp_ids); static struct snd_pci_quirk atiixp_quirks[] __devinitdata = { - SND_PCI_QUIRK(0x15bd, 0x3100, "DFI RS482", 0), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3100, "DFI RS482", 0), { } /* terminator */ };

2 1

[alsa-devel] [PATCH] hda: Add STAC92HD83XXX_PWR_REF quirk
by Matthew Ranostay 23 Jan '09

23 Jan '09

Some revisions of the 92hd8xxx codec's not supporting port power downs in which the using of it causes capture and also randomly playback streams to not function at all. Thus by disabling it by default and adding a option to enable it manually will fix all issue on current and future revisions. Signed-by-off: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 75914bc..2119175 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -349,6 +349,7 @@ STAC92HD73* STAC92HD83* =========== ref Reference board + mic-ref Reference board with power managment for ports STAC9872 ======== diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index dbe8b12..156f6fa 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -81,6 +81,7 @@ enum { enum { STAC_92HD83XXX_REF, + STAC_92HD83XXX_PWR_REF, STAC_92HD83XXX_MODELS }; @@ -1734,10 +1735,12 @@ static unsigned int ref92hd83xxx_pin_configs[14] = { static unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_REF] = ref92hd83xxx_pin_configs, + [STAC_92HD83XXX_PWR_REF] = ref92hd83xxx_pin_configs, }; static const char *stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_REF] = "ref", + [STAC_92HD83XXX_PWR_REF] = "mic-ref", }; static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { @@ -4781,13 +4784,6 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) AC_VERB_SET_CONNECT_SEL, num_dacs); spec->init = stac92hd83xxx_core_init; - switch (codec->vendor_id) { - case 0x111d7605: - break; - default: - spec->num_pwrs--; - } - spec->mixer = stac92hd83xxx_mixer; spec->num_pins = ARRAY_SIZE(stac92hd83xxx_pin_nids); spec->num_dmuxes = ARRAY_SIZE(stac92hd83xxx_dmux_nids); @@ -4813,6 +4809,15 @@ again: return err; } + switch (codec->vendor_id) { + case 0x111d7604: + case 0x111d7605: + if (spec->board_config == STAC_92HD83XXX_PWR_REF) + break; + spec->num_pwrs = 0; + break; + } + err = stac92xx_parse_auto_config(codec, 0x1d, 0); if (!err) { if (spec->board_config < 0) {

2 1

[alsa-devel] [PATCH] hda: revert change to 92hd83xxx power mapping
by Matthew Ranostay 23 Jan '09

23 Jan '09

Port 0xe power mapping was incorrect set to 0x80 changed to the correct value 0x40. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index dbe8b12..dd750d6 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -334,7 +334,7 @@ static hda_nid_t stac92hd83xxx_slave_dig_outs[2] = { }; static unsigned int stac92hd83xxx_pwr_mapping[4] = { - 0x03, 0x0c, 0x20, 0x80, + 0x03, 0x0c, 0x20, 0x40, }; static hda_nid_t stac92hd83xxx_amp_nids[1] = {

2 1

[alsa-devel] [PATCH 2.6.30] ASoC: remove stand-alone mode support from CS4270 codec driver
by Timur Tabi 23 Jan '09

23 Jan '09

The CS4270 supports stand-alone mode, where the codec is not connect to the I2C or SPI buses. Instead, input voltages configure the codec at power-on. The CS4270 ASoC device driver has partial support for this mode, but the code was never tested, and partial support doesn't help anyone. It also made the rest of the code more complicated than necessary. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/codecs/cs4270.c | 92 +++++++++++++------------------------------- 1 files changed, 27 insertions(+), 65 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index f1aa0c3..2e4ce04 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -12,11 +12,7 @@ * * Current features/limitations: * - * 1) Software mode is supported. Stand-alone mode is automatically - * selected if I2C is disabled or if a CS4270 is not found on the I2C - * bus. However, stand-alone mode is only partially implemented because - * there is no mechanism yet for this driver and the machine driver to - * communicate the values of the M0, M1, MCLK1, and MCLK2 pins. + * 1) Software mode is supported. Stand-alone mode is not supported. * 2) Only I2C is supported, not SPI * 3) Only Master mode is supported, not Slave. * 4) The machine driver's 'startup' function must call @@ -33,14 +29,6 @@ #include <sound/initval.h> #include <linux/i2c.h> -#include "cs4270.h" - -/* If I2C is defined, then we support software mode. However, if we're - not compiled as module but I2C is, then we can't use I2C calls. */ -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -#define USE_I2C -#endif - /* Private data for the CS4270 */ struct cs4270_private { unsigned int mclk; /* Input frequency of the MCLK pin */ @@ -60,8 +48,6 @@ struct cs4270_private { SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE) -#ifdef USE_I2C - /* CS4270 registers addresses */ #define CS4270_CHIPID 0x01 /* Chip ID */ #define CS4270_PWRCTL 0x02 /* Power Control */ @@ -272,17 +258,6 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, } /* - * A list of addresses on which this CS4270 could use. I2C addresses are - * 7 bits. For the CS4270, the upper four bits are always 1001, and the - * lower three bits are determined via the AD2, AD1, and AD0 pins - * (respectively). - */ -static const unsigned short normal_i2c[] = { - 0x48, 0x49, 0x4A, 0x4B, 0x4C, 0x4D, 0x4E, 0x4F, I2C_CLIENT_END -}; -I2C_CLIENT_INSMOD; - -/* * Pre-fill the CS4270 register cache. * * We use the auto-increment feature of the CS4270 to read all registers in @@ -476,7 +451,6 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, } #ifdef CONFIG_SND_SOC_CS4270_HWMUTE - /* * Set the CS4270 external mute * @@ -501,32 +475,16 @@ static int cs4270_mute(struct snd_soc_dai *dai, int mute) return snd_soc_write(codec, CS4270_MUTE, reg6); } - +#else +#define cs4270_mute NULL #endif -static int cs4270_i2c_probe(struct i2c_client *, const struct i2c_device_id *); - /* A list of non-DAPM controls that the CS4270 supports */ static const struct snd_kcontrol_new cs4270_snd_controls[] = { SOC_DOUBLE_R("Master Playback Volume", CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 1) }; -static const struct i2c_device_id cs4270_id[] = { - {"cs4270", 0}, - {} -}; -MODULE_DEVICE_TABLE(i2c, cs4270_id); - -static struct i2c_driver cs4270_i2c_driver = { - .driver = { - .name = "CS4270 I2C", - .owner = THIS_MODULE, - }, - .id_table = cs4270_id, - .probe = cs4270_i2c_probe, -}; - /* * Global variable to store socdev for i2c probe function. * @@ -633,7 +591,20 @@ error: return ret; } -#endif /* USE_I2C*/ +static const struct i2c_device_id cs4270_id[] = { + {"cs4270", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs4270_id); + +static struct i2c_driver cs4270_i2c_driver = { + .driver = { + .name = "cs4270", + .owner = THIS_MODULE, + }, + .id_table = cs4270_id, + .probe = cs4270_i2c_probe, +}; struct snd_soc_dai cs4270_dai = { .name = "CS4270", @@ -698,7 +669,6 @@ static int cs4270_probe(struct platform_device *pdev) goto error_free_codec; } -#ifdef USE_I2C cs4270_socdev = socdev; ret = i2c_add_driver(&cs4270_i2c_driver); @@ -708,20 +678,16 @@ static int cs4270_probe(struct platform_device *pdev) } /* Did we find a CS4270 on the I2C bus? */ - if (codec->control_data) { - /* Initialize codec ops */ - cs4270_dai.ops.hw_params = cs4270_hw_params; - cs4270_dai.ops.set_sysclk = cs4270_set_dai_sysclk; - cs4270_dai.ops.set_fmt = cs4270_set_dai_fmt; -#ifdef CONFIG_SND_SOC_CS4270_HWMUTE - cs4270_dai.ops.digital_mute = cs4270_mute; -#endif - } else - printk(KERN_INFO "cs4270: no I2C device found, " - "using stand-alone mode\n"); -#else - printk(KERN_INFO "cs4270: I2C disabled, using stand-alone mode\n"); -#endif + if (!codec->control_data) { + printk(KERN_ERR "cs4270: failed to attach driver"); + goto error_del_driver; + } + + /* Initialize codec ops */ + cs4270_dai.ops.hw_params = cs4270_hw_params; + cs4270_dai.ops.set_sysclk = cs4270_set_dai_sysclk; + cs4270_dai.ops.set_fmt = cs4270_set_dai_fmt; + cs4270_dai.ops.digital_mute = cs4270_mute; ret = snd_soc_init_card(socdev); if (ret < 0) { @@ -732,11 +698,9 @@ static int cs4270_probe(struct platform_device *pdev) return 0; error_del_driver: -#ifdef USE_I2C i2c_del_driver(&cs4270_i2c_driver); error_free_pcms: -#endif snd_soc_free_pcms(socdev); error_free_codec: @@ -752,9 +716,7 @@ static int cs4270_remove(struct platform_device *pdev) snd_soc_free_pcms(socdev); -#ifdef USE_I2C i2c_del_driver(&cs4270_i2c_driver); -#endif kfree(socdev->codec); socdev->codec = NULL; -- 1.5.5

1 0

[alsa-devel] Moved next/* to topic/* branches on sound git tree
by Takashi Iwai 22 Jan '09

22 Jan '09

Hi, right now I moved all next/* branches into topic/* branches on my sound GIT tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git All these branches have been merged into for-next branch so that it'll be included in the next tree from tomorrow. That is, topic/* branches are opened for 2.6.30 kernel development. The fixes for 2.6.29 kernel are put into fix/* branches. If your local tree is referring to the old branches, please update to the new branches appropriately. thanks, Takashi

1 0

[alsa-devel] [PATCH] hda: Add STAC92HD83XXX_PWR_REF quirk
by Matthew Ranostay 22 Jan '09

22 Jan '09

Some revisions of the 92hd8xxx codec's not supporting port power downs in which the using of it causes capture and also randomly playback streams to not function at all. Thus by disabling it by default and adding a option to enable it manually will fix all issue on current and future revisions. Signed-by-off: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 75914bc..2119175 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -349,6 +349,7 @@ STAC92HD73* STAC92HD83* =========== ref Reference board + mic-ref Reference board with power managment for ports STAC9872 ======== diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index dbe8b12..a961c51 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -81,6 +81,7 @@ enum { enum { STAC_92HD83XXX_REF, + STAC_92HD83XXX_PWR_REF, STAC_92HD83XXX_MODELS }; @@ -334,7 +335,7 @@ static hda_nid_t stac92hd83xxx_slave_dig_outs[2] = { }; static unsigned int stac92hd83xxx_pwr_mapping[4] = { - 0x03, 0x0c, 0x20, 0x80, + 0x03, 0x0c, 0x20, 0x40, }; static hda_nid_t stac92hd83xxx_amp_nids[1] = { @@ -1734,10 +1735,12 @@ static unsigned int ref92hd83xxx_pin_configs[14] = { static unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_REF] = ref92hd83xxx_pin_configs, + [STAC_92HD83XXX_PWR_REF] = ref92hd83xxx_pin_configs, }; static const char *stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_REF] = "ref", + [STAC_92HD83XXX_PWR_REF] = "mic-ref", }; static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { @@ -4781,13 +4784,6 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) AC_VERB_SET_CONNECT_SEL, num_dacs); spec->init = stac92hd83xxx_core_init; - switch (codec->vendor_id) { - case 0x111d7605: - break; - default: - spec->num_pwrs--; - } - spec->mixer = stac92hd83xxx_mixer; spec->num_pins = ARRAY_SIZE(stac92hd83xxx_pin_nids); spec->num_dmuxes = ARRAY_SIZE(stac92hd83xxx_dmux_nids); @@ -4813,6 +4809,15 @@ again: return err; } + switch (codec->vendor_id) { + case 0x111d7604: + case 0x111d7605: + if (spec->board_config == STAC_92HD83XXX_PWR_REF) + break; + spec->num_pwrs = 0; + break; + } + err = stac92xx_parse_auto_config(codec, 0x1d, 0); if (!err) { if (spec->board_config < 0) {

2 3

[alsa-devel] Independent usage of two channels
by Pavel Fedin 22 Jan '09

22 Jan '09

Hello! I'm novice in ALSA programming and i've got a question. Is it possible to use two channels of a sound card as two independent devices for PCM playback? I'm making a PC-based security system, and i would like to use one channel as a voice feedback for control panel and another one - as a software-driven siren. Kind regards

2 1

[alsa-devel] [PATCH 1/2] New Control Plugin - arcam-av
by Peter Stokes 22 Jan '09

22 Jan '09

Signed-off-by: Peter Stokes <linux(a)dadeos.co.uk> diff -Nrup alsa-plugins-1.0.18-orig/Makefile.am alsa-plugins-1.0.18/Makefile.am --- alsa-plugins-1.0.18-orig/Makefile.am 2008-10-29 12:42:13.000000000 +0000 +++ alsa-plugins-1.0.18/Makefile.am 2009-01-02 22:22:26.000000000 +0000 @@ -18,7 +18,7 @@ if HAVE_PPH PPHDIR = pph endif -SUBDIRS = oss mix $(PPHDIR) $(JACKDIR) $(PULSEDIR) $(SAMPLERATEDIR) $(A52DIR) $(LAVCRATEDIR) $(MAEMODIR) usb_stream doc +SUBDIRS = oss mix $(PPHDIR) $(JACKDIR) $(PULSEDIR) $(SAMPLERATEDIR) $(A52DIR) $(LAVCRATEDIR) $(MAEMODIR) usb_stream arcam-av doc EXTRA_DIST = gitcompile version COPYING.GPL AUTOMAKE_OPTIONS = foreign diff -Nrup alsa-plugins-1.0.18-orig/Makefile.in alsa-plugins-1.0.18/Makefile.in --- alsa-plugins-1.0.18-orig/Makefile.in 2008-10-29 12:47:44.000000000 +0000 +++ alsa-plugins-1.0.18/Makefile.in 2009-01-02 22:22:47.000000000 +0000 @@ -61,7 +61,7 @@ RECURSIVE_TARGETS = all-recursive check- ETAGS = etags CTAGS = ctags DIST_SUBDIRS = oss mix pph jack pulse rate a52 rate-lavc maemo \ - usb_stream doc + usb_stream arcam-av doc DISTFILES = $(DIST_COMMON) $(DIST_SOURCES) $(TEXINFOS) $(EXTRA_DIST) distdir = $(PACKAGE)-$(VERSION) top_distdir = $(distdir) @@ -212,7 +212,7 @@ target_alias = @target_alias@ @HAVE_AVCODEC_TRUE@LAVCRATEDIR = rate-lavc @HAVE_MAEMO_PLUGIN_TRUE@MAEMODIR = maemo @HAVE_PPH_TRUE@PPHDIR = pph -SUBDIRS = oss mix $(PPHDIR) $(JACKDIR) $(PULSEDIR) $(SAMPLERATEDIR) $(A52DIR) $(LAVCRATEDIR) $(MAEMODIR) usb_stream doc +SUBDIRS = oss mix $(PPHDIR) $(JACKDIR) $(PULSEDIR) $(SAMPLERATEDIR) $(A52DIR) $(LAVCRATEDIR) $(MAEMODIR) usb_stream arcam-av doc EXTRA_DIST = gitcompile version COPYING.GPL AUTOMAKE_OPTIONS = foreign all: config.h diff -Nrup alsa-plugins-1.0.18-orig/arcam-av/Makefile.am alsa-plugins-1.0.18/arcam-av/Makefile.am --- alsa-plugins-1.0.18-orig/arcam-av/Makefile.am 1970-01-01 01:00:00.000000000 +0100 +++ alsa-plugins-1.0.18/arcam-av/Makefile.am 2009-01-02 22:31:03.000000000 +0000 @@ -0,0 +1,9 @@ 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3 5

[alsa-devel] [PATCH] Turtle Beach Multisound Classic/Pinnacle driver (2nd rev)
by Krzysztof Helt 22 Jan '09

22 Jan '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This is driver for Turtle Beach Multisound cards: Classic, Fiji and Pinnacle. Tested pcm playback and recording and MIDI playback on Multisound Pinnacle. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The list of improvements since the previously posted driver: - CS clean up - fixed 8-bit sample audio playback and recording - fixed recording formats - correctly request shared memory region - ioremap memory region once instead before each access - single module per driver/removed common module - reduced midi code as it was copied from mpu401 driver and is redundant now - various clean ups and code reductions Not tested: 1. Classic card (no hardware) 2. Classic card PnP detection (tested only on the Pinnacle) (no hardware) 3. Digital input/output 4. External MIDI input/output I would be grateful for testing and reporting any of these. Best regards, Krzysztof diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 660beb4..2d6ce32 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -411,5 +411,36 @@ config SND_WAVEFRONT_FIRMWARE_IN_KERNEL you need to install the firmware files from the alsa-firmware package. +config SND_MSND_PINNACLE + tristate "Turtle Beach MultiSound Pinnacle/Fiji driver" + depends on X86 + select FW_LOADER + select SND_MPU401_UART + select SND_PCM + help + Say Y to include support for Turtle Beach MultiSound Pinnacle/ + Fiji soundcards. + + To compile this driver as a module, choose M here: the module + will be called snd-msnd-pinnacle. + +config SND_MSND_CLASSIC + tristate "Support for Turtle Beach MultiSound Classic, Tahiti, Monterey" + depends on X86 + select FW_LOADER + select SND_MPU401_UART + select SND_PCM + help + Say M here if you have a Turtle Beach MultiSound Classic, Tahiti or + Monterey (not for the Pinnacle or Fiji). + + See <file:Documentation/sound/oss/MultiSound> for important information + about this driver. Note that it has been discontinued, but the + Voyetra Turtle Beach knowledge base entry for it is still available + at <http://www.turtlebeach.com/site/kb_ftp/790.asp>. + + To compile this driver as a module, choose M here: the module + will be called snd-msnd-classic. + endif # SND_ISA diff --git a/sound/isa/Makefile b/sound/isa/Makefile index 63af13d..beb96ec 100644 --- a/sound/isa/Makefile +++ b/sound/isa/Makefile @@ -26,5 +26,5 @@ obj-$(CONFIG_SND_SC6000) += snd-sc6000.o obj-$(CONFIG_SND_SGALAXY) += snd-sgalaxy.o obj-$(CONFIG_SND_SSCAPE) += snd-sscape.o -obj-$(CONFIG_SND) += ad1816a/ ad1848/ cs423x/ es1688/ gus/ opti9xx/ \ - sb/ wavefront/ wss/ +obj-$(CONFIG_SND) += ad1816a/ ad1848/ cs423x/ es1688/ gus/ msnd/ \ + opti9xx/ sb/ wavefront/ wss/ diff --git a/sound/isa/msnd/Makefile b/sound/isa/msnd/Makefile new file mode 100644 index 0000000..41f278f --- /dev/null +++ b/sound/isa/msnd/Makefile @@ -0,0 +1,10 @@ + +snd-msnd-pinnacle-objs := msnd_pinnacle.o msnd_pinnacle_mixer.o \ + msnd.o msnd_midi.o +snd-msnd-classic-objs := msnd_classic.o msnd_pinnacle_mixer.o \ + msnd.o msnd_midi.o + +# Toplevel Module Dependency +obj-$(CONFIG_SND_MSND_PINNACLE) += snd-msnd-pinnacle.o +obj-$(CONFIG_SND_MSND_CLASSIC) += snd-msnd-classic.o + diff --git a/sound/isa/msnd/msnd.c b/sound/isa/msnd/msnd.c new file mode 100644 index 0000000..5326834 --- /dev/null +++ b/sound/isa/msnd/msnd.c @@ -0,0 +1,192 @@ +/********************************************************************* + * + * 2002/06/30 Karsten Wiese: + * removed kernel-version dependencies. + * ripped from linux kernel 2.4.18 (OSS Implementation) by me. + * In the OSS Version, this file is compiled to a separate MODULE, + * that is used by the pinnacle and the classic driver. + * since there is no classic driver for alsa yet (i dont have a classic + * & writing one blindfold is difficult) this file's object is statically + * linked into the pinnacle-driver-module for now. look for the string + * "uncomment this to make this a module again" + * to do guess what. + * + * the following is a copy of the 2.4.18 OSS FREE file-heading comment: + * + * msnd.c - Driver Base + * + * Turtle Beach MultiSound Sound Card Driver for Linux + * + * Copyright (C) 1998 Andrew Veliath + * + * 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., 675 Mass Ave, Cambridge, MA 02139, USA. + * + ********************************************************************/ + +#include <sound/core.h> +#include <sound/initval.h> +#include <linux/kernel.h> +#include <linux/types.h> +#include <linux/interrupt.h> +#include <linux/io.h> + +#include "msnd.h" + +#define LOGNAME "msnd" + + +void snd_msnd_init_queue(void *base, int start, int size) +{ + writew(PCTODSP_BASED(start), base + JQS_wStart); + writew(PCTODSP_OFFSET(size) - 1, base + JQS_wSize); + writew(0, base + JQS_wHead); + writew(0, base + JQS_wTail); +} + +static int snd_msnd_wait_TXDE(struct snd_msnd *dev) +{ + unsigned int io = dev->io; + int timeout = 1000; + + while (timeout-- > 0) + if (inb(io + HP_ISR) & HPISR_TXDE) + return 0; + + return -EIO; +} + +static int snd_msnd_wait_HC0(struct snd_msnd *dev) +{ + unsigned int io = dev->io; + int timeout = 1000; + + while (timeout-- > 0) + if (!(inb(io + HP_CVR) & HPCVR_HC)) + return 0; + + return -EIO; +} + +int snd_msnd_send_dsp_cmd(struct snd_msnd *dev, u8 cmd) +{ + unsigned long flags; + + spin_lock_irqsave(&dev->lock, flags); + if (snd_msnd_wait_HC0(dev) == 0) { + outb(cmd, dev->io + HP_CVR); + spin_unlock_irqrestore(&dev->lock, flags); + return 0; + } + spin_unlock_irqrestore(&dev->lock, flags); + + snd_printd(LOGNAME ": Send DSP command timeout\n"); + + return -EIO; +} + +int snd_msnd_send_word(struct snd_msnd *dev, unsigned char high, + unsigned char mid, unsigned char low) +{ + unsigned int io = dev->io; + + if (snd_msnd_wait_TXDE(dev) == 0) { + outb(high, io + HP_TXH); + outb(mid, io + HP_TXM); + outb(low, io + HP_TXL); + return 0; + } + + snd_printd(LOGNAME ": Send host word timeout\n"); + + return -EIO; +} + +int snd_msnd_upload_host(struct snd_msnd *dev, const u8 *bin, int len) +{ + int i; + + if (len % 3 != 0) { + snd_printk(LOGNAME ": Upload host data not multiple of 3!\n"); + return -EINVAL; + } + + for (i = 0; i < len; i += 3) + if (snd_msnd_send_word(dev, bin[i], bin[i + 1], bin[i + 2])) + return -EIO; + + inb(dev->io + HP_RXL); + inb(dev->io + HP_CVR); + + return 0; +} + +int snd_msnd_enable_irq(struct snd_msnd *dev) +{ + unsigned long flags; + + if (dev->irq_ref++) + return 0; + + snd_printd(LOGNAME ": Enabling IRQ\n"); + + spin_lock_irqsave(&dev->lock, flags); + if (snd_msnd_wait_TXDE(dev) == 0) { + outb(inb(dev->io + HP_ICR) | HPICR_TREQ, dev->io + HP_ICR); + if (dev->type == msndClassic) + outb(dev->irqid, dev->io + HP_IRQM); + + outb(inb(dev->io + HP_ICR) & ~HPICR_TREQ, dev->io + HP_ICR); + outb(inb(dev->io + HP_ICR) | HPICR_RREQ, dev->io + HP_ICR); + enable_irq(dev->irq); + snd_msnd_init_queue(dev->DSPQ, dev->dspq_data_buff, + dev->dspq_buff_size); + spin_unlock_irqrestore(&dev->lock, flags); + return 0; + } + spin_unlock_irqrestore(&dev->lock, flags); + + snd_printd(LOGNAME ": Enable IRQ failed\n"); + + return -EIO; +} + +int snd_msnd_disable_irq(struct snd_msnd *dev) +{ + unsigned long flags; + + if (--dev->irq_ref > 0) + return 0; + + if (dev->irq_ref < 0) + snd_printd(LOGNAME ": IRQ ref count is %d\n", dev->irq_ref); + + snd_printd(LOGNAME ": Disabling IRQ\n"); + + spin_lock_irqsave(&dev->lock, flags); + if (snd_msnd_wait_TXDE(dev) == 0) { + outb(inb(dev->io + HP_ICR) & ~HPICR_RREQ, dev->io + HP_ICR); + if (dev->type == msndClassic) + outb(HPIRQ_NONE, dev->io + HP_IRQM); + disable_irq(dev->irq); + spin_unlock_irqrestore(&dev->lock, flags); + return 0; + } + spin_unlock_irqrestore(&dev->lock, flags); + + snd_printd(LOGNAME ": Disable IRQ failed\n"); + + return -EIO; +} + diff --git a/sound/isa/msnd/msnd.h b/sound/isa/msnd/msnd.h new file mode 100644 index 0000000..018b672 --- /dev/null +++ b/sound/isa/msnd/msnd.h @@ -0,0 +1,293 @@ +/********************************************************************* + * + * msnd.h + * + * Turtle Beach MultiSound Sound Card Driver for Linux + * + * Some parts of this header file were derived from the Turtle Beach + * MultiSound Driver Development Kit. + * + * Copyright (C) 1998 Andrew Veliath + * Copyright (C) 1993 Turtle Beach Systems, Inc. + * + * 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., 675 Mass Ave, Cambridge, MA 02139, USA. + * + ********************************************************************/ +#ifndef __MSND_H +#define __MSND_H + +#define DEFSAMPLERATE 44100 +#define DEFSAMPLESIZE SNDRV_PCM_FORMAT_S16 +#define DEFCHANNELS 1 + +#define SRAM_BANK_SIZE 0x8000 +#define SRAM_CNTL_START 0x7F00 +#define SMA_STRUCT_START 0x7F40 + +#define DSP_BASE_ADDR 0x4000 +#define DSP_BANK_BASE 0x4000 + +#define AGND 0x01 +#define SIGNAL 0x02 + +#define EXT_DSP_BIT_DCAL 0x0001 +#define EXT_DSP_BIT_MIDI_CON 0x0002 + +#define BUFFSIZE 0x8000 +#define HOSTQ_SIZE 0x40 + +#define DAP_BUFF_SIZE 0x2400 + +#define DAPQ_STRUCT_SIZE 0x10 +#define DARQ_STRUCT_SIZE 0x10 +#define DAPQ_BUFF_SIZE (3 * 0x10) +#define DARQ_BUFF_SIZE (3 * 0x10) +#define MODQ_BUFF_SIZE 0x400 + +#define DAPQ_DATA_BUFF 0x6C00 +#define DARQ_DATA_BUFF 0x6C30 +#define MODQ_DATA_BUFF 0x6C60 +#define MIDQ_DATA_BUFF 0x7060 + +#define DAPQ_OFFSET SRAM_CNTL_START +#define DARQ_OFFSET (SRAM_CNTL_START + 0x08) +#define MODQ_OFFSET (SRAM_CNTL_START + 0x10) +#define MIDQ_OFFSET (SRAM_CNTL_START + 0x18) +#define DSPQ_OFFSET (SRAM_CNTL_START + 0x20) + +#define HP_ICR 0x00 +#define HP_CVR 0x01 +#define HP_ISR 0x02 +#define HP_IVR 0x03 +#define HP_NU 0x04 +#define HP_INFO 0x04 +#define HP_TXH 0x05 +#define HP_RXH 0x05 +#define HP_TXM 0x06 +#define HP_RXM 0x06 +#define HP_TXL 0x07 +#define HP_RXL 0x07 + +#define HP_ICR_DEF 0x00 +#define HP_CVR_DEF 0x12 +#define HP_ISR_DEF 0x06 +#define HP_IVR_DEF 0x0f +#define HP_NU_DEF 0x00 + +#define HP_IRQM 0x09 + +#define HPR_BLRC 0x08 +#define HPR_SPR1 0x09 +#define HPR_SPR2 0x0A +#define HPR_TCL0 0x0B +#define HPR_TCL1 0x0C +#define HPR_TCL2 0x0D +#define HPR_TCL3 0x0E +#define HPR_TCL4 0x0F + +#define HPICR_INIT 0x80 +#define HPICR_HM1 0x40 +#define HPICR_HM0 0x20 +#define HPICR_HF1 0x10 +#define HPICR_HF0 0x08 +#define HPICR_TREQ 0x02 +#define HPICR_RREQ 0x01 + +#define HPCVR_HC 0x80 + +#define HPISR_HREQ 0x80 +#define HPISR_DMA 0x40 +#define HPISR_HF3 0x10 +#define HPISR_HF2 0x08 +#define HPISR_TRDY 0x04 +#define HPISR_TXDE 0x02 +#define HPISR_RXDF 0x01 + +#define HPIO_290 0 +#define HPIO_260 1 +#define HPIO_250 2 +#define HPIO_240 3 +#define HPIO_230 4 +#define HPIO_220 5 +#define HPIO_210 6 +#define HPIO_3E0 7 + +#define HPMEM_NONE 0 +#define HPMEM_B000 1 +#define HPMEM_C800 2 +#define HPMEM_D000 3 +#define HPMEM_D400 4 +#define HPMEM_D800 5 +#define HPMEM_E000 6 +#define HPMEM_E800 7 + +#define HPIRQ_NONE 0 +#define HPIRQ_5 1 +#define HPIRQ_7 2 +#define HPIRQ_9 3 +#define HPIRQ_10 4 +#define HPIRQ_11 5 +#define HPIRQ_12 6 +#define HPIRQ_15 7 + +#define HIMT_PLAY_DONE 0x00 +#define HIMT_RECORD_DONE 0x01 +#define HIMT_MIDI_EOS 0x02 +#define HIMT_MIDI_OUT 0x03 + +#define HIMT_MIDI_IN_UCHAR 0x0E +#define HIMT_DSP 0x0F + +#define HDEX_BASE 0x92 +#define HDEX_PLAY_START (0 + HDEX_BASE) +#define HDEX_PLAY_STOP (1 + HDEX_BASE) +#define HDEX_PLAY_PAUSE (2 + HDEX_BASE) +#define HDEX_PLAY_RESUME (3 + HDEX_BASE) +#define HDEX_RECORD_START (4 + HDEX_BASE) +#define HDEX_RECORD_STOP (5 + HDEX_BASE) +#define HDEX_MIDI_IN_START (6 + HDEX_BASE) +#define HDEX_MIDI_IN_STOP (7 + HDEX_BASE) +#define HDEX_MIDI_OUT_START (8 + HDEX_BASE) +#define HDEX_MIDI_OUT_STOP (9 + HDEX_BASE) +#define HDEX_AUX_REQ (10 + HDEX_BASE) + +#define HIWORD(l) ((u16)((((u32)(l)) >> 16) & 0xFFFF)) +#define LOWORD(l) ((u16)(u32)(l)) +#define HIBYTE(w) ((u8)(((u16)(w) >> 8) & 0xFF)) +#define LOBYTE(w) ((u8)(w)) +#define MAKELONG(low, hi) ((long)(((u16)(low))|(((u32)((u16)(hi)))<<16))) +#define MAKEWORD(low, hi) ((u16)(((u8)(low))|(((u16)((u8)(hi)))<<8))) + +#define PCTODSP_OFFSET(w) (u16)((w)/2) +#define PCTODSP_BASED(w) (u16)(((w)/2) + DSP_BASE_ADDR) +#define DSPTOPC_BASED(w) (((w) - DSP_BASE_ADDR) * 2) + +#ifdef SLOWIO +# undef outb +# undef inb +# define outb outb_p +# define inb inb_p +#endif + +/* JobQueueStruct */ +#define JQS_wStart 0x00 +#define JQS_wSize 0x02 +#define JQS_wHead 0x04 +#define JQS_wTail 0x06 +#define JQS__size 0x08 + +/* DAQueueDataStruct */ +#define DAQDS_wStart 0x00 +#define DAQDS_wSize 0x02 +#define DAQDS_wFormat 0x04 +#define DAQDS_wSampleSize 0x06 +#define DAQDS_wChannels 0x08 +#define DAQDS_wSampleRate 0x0A +#define DAQDS_wIntMsg 0x0C +#define DAQDS_wFlags 0x0E +#define DAQDS__size 0x10 + +/* Generic FIFO * / +typedef struct { + size_t n, len; + char *data; + int head, tail; +} msnd_fifo; */ + + + +struct snd_msnd { + void *mappedbase; + int play_period_bytes; + int playLimit; + int playPeriods; + int playDMAPos; + int banksPlayed; + int captureDMAPos; + int capturePeriodBytes; + int captureLimit; + int capturePeriods; + struct snd_card *card; + void *msndmidi_mpu; + struct snd_rawmidi *rmidi; + + /* Hardware resources */ + long io; + int memid, irqid; + int irq, irq_ref; + unsigned long base; + + /* Motorola 56k DSP SMA */ + void *SMA; + void *DAPQ; + void *DARQ; + void *MODQ; + void *MIDQ; + void *DSPQ; + int dspq_data_buff, dspq_buff_size; + + /* State variables */ + enum { msndClassic, msndPinnacle } type; + mode_t mode; + unsigned long flags; +#define F_RESETTING 0 +#define F_HAVEDIGITAL 1 +#define F_AUDIO_WRITE_INUSE 2 +#define F_WRITING 3 +#define F_WRITEBLOCK 4 +#define F_WRITEFLUSH 5 +#define F_AUDIO_READ_INUSE 6 +#define F_READING 7 +#define F_READBLOCK 8 +#define F_EXT_MIDI_INUSE 9 +#define F_HDR_MIDI_INUSE 10 +#define F_DISABLE_WRITE_NDELAY 11 + spinlock_t lock; + spinlock_t mixer_lock; + int nresets; + unsigned recsrc; +#define LEVEL_ENTRIES 32 + int left_levels[LEVEL_ENTRIES]; + int right_levels[LEVEL_ENTRIES]; + int calibrate_signal; + int play_sample_size, play_sample_rate, play_channels; + int play_ndelay; + int capture_sample_size, capture_sample_rate, capture_channels; + int capture_ndelay; + u8 bCurrentMidiPatch; + + int last_playbank, last_recbank; + struct snd_pcm_substream *playback_substream; + struct snd_pcm_substream *capture_substream; + +}; + +void snd_msnd_init_queue(void *base, int start, int size); + +int snd_msnd_send_dsp_cmd(struct snd_msnd *chip, u8 cmd); +int snd_msnd_send_word(struct snd_msnd *chip, + unsigned char high, + unsigned char mid, + unsigned char low); +int snd_msnd_upload_host(struct snd_msnd *chip, + const u8 *bin, int len); +int snd_msnd_enable_irq(struct snd_msnd *chip); +int snd_msnd_disable_irq(struct snd_msnd *chip); + + +int snd_msndmidi_new(struct snd_card *card, int device); +void snd_msndmidi_input_read(void *mpu); + +#endif /* __MSND_H */ diff --git a/sound/isa/msnd/msnd_classic.c b/sound/isa/msnd/msnd_classic.c new file mode 100644 index 0000000..3b23a09 --- /dev/null +++ b/sound/isa/msnd/msnd_classic.c @@ -0,0 +1,3 @@ +/* The work is in msnd_pinnacle.c, just define MSND_CLASSIC before it. */ +#define MSND_CLASSIC +#include "msnd_pinnacle.c" diff --git a/sound/isa/msnd/msnd_classic.h b/sound/isa/msnd/msnd_classic.h new file mode 100644 index 0000000..48cce62 --- /dev/null +++ b/sound/isa/msnd/msnd_classic.h @@ -0,0 +1,159 @@ +/********************************************************************* + * + * msnd_classic.h + * + * Turtle Beach MultiSound Sound Card Driver for Linux + * + * Some parts of this header file were derived from the Turtle Beach + * MultiSound Driver Development Kit. + * + * Copyright (C) 1998 Andrew Veliath + * Copyright (C) 1993 Turtle Beach Systems, Inc. + * + * 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., 675 Mass Ave, Cambridge, MA 02139, USA. + * + ********************************************************************/ +#ifndef __MSND_CLASSIC_H +#define __MSND_CLASSIC_H + +#define DSP_NUMIO 0x10 + +#define HP_MEMM 0x08 + +#define HP_BITM 0x0E +#define HP_WAIT 0x0D +#define HP_DSPR 0x0A +#define HP_PROR 0x0B +#define HP_BLKS 0x0C + +#define HPPRORESET_OFF 0 +#define HPPRORESET_ON 1 + +#define HPDSPRESET_OFF 0 +#define HPDSPRESET_ON 1 + +#define HPBLKSEL_0 0 +#define HPBLKSEL_1 1 + +#define HPWAITSTATE_0 0 +#define HPWAITSTATE_1 1 + +#define HPBITMODE_16 0 +#define HPBITMODE_8 1 + +#define HIDSP_INT_PLAY_UNDER 0x00 +#define HIDSP_INT_RECORD_OVER 0x01 +#define HIDSP_INPUT_CLIPPING 0x02 +#define HIDSP_MIDI_IN_OVER 0x10 +#define HIDSP_MIDI_OVERRUN_ERR 0x13 + +#define HDEXAR_CLEAR_PEAKS 1 +#define HDEXAR_IN_SET_POTS 2 +#define HDEXAR_AUX_SET_POTS 3 +#define HDEXAR_CAL_A_TO_D 4 +#define HDEXAR_RD_EXT_DSP_BITS 5 + +#define TIME_PRO_RESET_DONE 0x028A +#define TIME_PRO_SYSEX 0x0040 +#define TIME_PRO_RESET 0x0032 + +#define DAR_BUFF_SIZE 0x2000 + +#define MIDQ_BUFF_SIZE 0x200 +#define DSPQ_BUFF_SIZE 0x40 + +#define DSPQ_DATA_BUFF 0x7260 + +#define MOP_SYNTH 0x10 +#define MOP_EXTOUT 0x32 +#define MOP_EXTTHRU 0x02 +#define MOP_OUTMASK 0x01 + +#define MIP_EXTIN 0x01 +#define MIP_SYNTH 0x00 +#define MIP_INMASK 0x32 + +/* Classic SMA Common Data */ +#define SMA_wCurrPlayBytes 0x0000 +#define SMA_wCurrRecordBytes 0x0002 +#define SMA_wCurrPlayVolLeft 0x0004 +#define SMA_wCurrPlayVolRight 0x0006 +#define SMA_wCurrInVolLeft 0x0008 +#define SMA_wCurrInVolRight 0x000a +#define SMA_wUser_3 0x000c +#define SMA_wUser_4 0x000e +#define SMA_dwUser_5 0x0010 +#define SMA_dwUser_6 0x0014 +#define SMA_wUser_7 0x0018 +#define SMA_wReserved_A 0x001a +#define SMA_wReserved_B 0x001c +#define SMA_wReserved_C 0x001e +#define SMA_wReserved_D 0x0020 +#define SMA_wReserved_E 0x0022 +#define SMA_wReserved_F 0x0024 +#define SMA_wReserved_G 0x0026 +#define SMA_wReserved_H 0x0028 +#define SMA_wCurrDSPStatusFlags 0x002a +#define SMA_wCurrHostStatusFlags 0x002c +#define SMA_wCurrInputTagBits 0x002e +#define SMA_wCurrLeftPeak 0x0030 +#define SMA_wCurrRightPeak 0x0032 +#define SMA_wExtDSPbits 0x0034 +#define SMA_bExtHostbits 0x0036 +#define SMA_bBoardLevel 0x0037 +#define SMA_bInPotPosRight 0x0038 +#define SMA_bInPotPosLeft 0x0039 +#define SMA_bAuxPotPosRight 0x003a +#define SMA_bAuxPotPosLeft 0x003b +#define SMA_wCurrMastVolLeft 0x003c +#define SMA_wCurrMastVolRight 0x003e +#define SMA_bUser_12 0x0040 +#define SMA_bUser_13 0x0041 +#define SMA_wUser_14 0x0042 +#define SMA_wUser_15 0x0044 +#define SMA_wCalFreqAtoD 0x0046 +#define SMA_wUser_16 0x0048 +#define SMA_wUser_17 0x004a +#define SMA__size 0x004c + +#ifdef HAVE_DSPCODEH +# include "msndperm.c" +# include "msndinit.c" +# define PERMCODE msndperm +# define INITCODE msndinit +# define PERMCODESIZE sizeof(msndperm) +# define INITCODESIZE sizeof(msndinit) +#else +# ifndef CONFIG_MSNDCLAS_INIT_FILE +# define CONFIG_MSNDCLAS_INIT_FILE "turtlebeach/msndinit.bin" +# endif +# ifndef CONFIG_MSNDCLAS_PERM_FILE +# define CONFIG_MSNDCLAS_PERM_FILE "turtlebeach/msndperm.bin" +# endif +# define PERMCODEFILE CONFIG_MSNDCLAS_PERM_FILE +# define INITCODEFILE CONFIG_MSNDCLAS_INIT_FILE +# define PERMCODE dspini +# define INITCODE permini +# define PERMCODESIZE sizeof_dspini +# define INITCODESIZE sizeof_permini +#endif +#define LONGNAME "MultiSound (Classic/Monterey/Tahiti)" + +/* functions from external mixer file */ +void snd_msndmix_setup(struct snd_msnd *chip); +unsigned long snd_msndmix_force_recsrc(struct snd_msnd *chip, int recsrc); +int __devinit snd_msndmix_new(struct snd_card *card); + +#endif /* __MSND_CLASSIC_H */ diff --git a/sound/isa/msnd/msnd_midi.c b/sound/isa/msnd/msnd_midi.c new file mode 100644 index 0000000..5f633c1 --- /dev/null +++ b/sound/isa/msnd/msnd_midi.c @@ -0,0 +1,178 @@ +/* + * Copyright (c) by Jaroslav Kysela <perex(a)perex.cz> + * Routines for control of MPU-401 in UART mode + * + * MPU-401 supports UART mode which is not capable generate transmit + * interrupts thus output is done via polling. Also, if irq < 0, then + * input is done also via polling. Do not expect good performance. + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <linux/io.h> +#include <linux/delay.h> +#include <linux/ioport.h> +#include <linux/errno.h> +#include <sound/core.h> +#include <sound/rawmidi.h> + +#include "msnd.h" + +#define MSNDMIDI_MODE_BIT_INPUT 0 +#define MSNDMIDI_MODE_BIT_OUTPUT 1 +#define MSNDMIDI_MODE_BIT_INPUT_TRIGGER 2 +#define MSNDMIDI_MODE_BIT_OUTPUT_TRIGGER 3 + +struct snd_msndmidi { + struct snd_msnd *dev; + + unsigned long mode; /* MSNDMIDI_MODE_XXXX */ + + struct snd_rawmidi_substream *substream_input; + + spinlock_t input_lock; +}; + +/* + * input/output open/close - protected by open_mutex in rawmidi.c + */ +static int snd_msndmidi_input_open(struct snd_rawmidi_substream *substream) +{ + struct snd_msndmidi *mpu; + + snd_printdd("snd_msndmidi_input_open()\n"); + + mpu = substream->rmidi->private_data; + + mpu->substream_input = substream; + + snd_msnd_enable_irq(mpu->dev); + + snd_msnd_send_dsp_cmd(mpu->dev, HDEX_MIDI_IN_START); + set_bit(MSNDMIDI_MODE_BIT_INPUT, &mpu->mode); + return 0; +} + +static int snd_msndmidi_input_close(struct snd_rawmidi_substream *substream) +{ + struct snd_msndmidi *mpu; + + mpu = substream->rmidi->private_data; + snd_msnd_send_dsp_cmd(mpu->dev, HDEX_MIDI_IN_STOP); + clear_bit(MSNDMIDI_MODE_BIT_INPUT, &mpu->mode); + mpu->substream_input = NULL; + snd_msnd_disable_irq(mpu->dev); + return 0; +} + +static void snd_msndmidi_input_drop(struct snd_msndmidi *mpu) +{ + u16 tail; + + tail = readw(mpu->dev->MIDQ + JQS_wTail); + writew(tail, mpu->dev->MIDQ + JQS_wHead); +} + +/* + * trigger input + */ +static void snd_msndmidi_input_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + unsigned long flags; + struct snd_msndmidi *mpu; + + snd_printdd("snd_msndmidi_input_trigger(, %i)\n", up); + + mpu = substream->rmidi->private_data; + spin_lock_irqsave(&mpu->input_lock, flags); + if (up) { + if (!test_and_set_bit(MSNDMIDI_MODE_BIT_INPUT_TRIGGER, + &mpu->mode)) + snd_msndmidi_input_drop(mpu); + } else { + clear_bit(MSNDMIDI_MODE_BIT_INPUT_TRIGGER, &mpu->mode); + } + spin_unlock_irqrestore(&mpu->input_lock, flags); + if (up) + snd_msndmidi_input_read(mpu); +} + +void snd_msndmidi_input_read(void *mpuv) +{ + unsigned long flags; + struct snd_msndmidi *mpu = mpuv; + void *pwMIDQData = mpu->dev->mappedbase + MIDQ_DATA_BUFF; + + spin_lock_irqsave(&mpu->input_lock, flags); + while (readw(mpu->dev->MIDQ + JQS_wTail) != + readw(mpu->dev->MIDQ + JQS_wHead)) { + u16 wTmp, val; + val = readw(pwMIDQData + 2 * readw(mpu->dev->MIDQ + JQS_wHead)); + + if (test_bit(MSNDMIDI_MODE_BIT_INPUT_TRIGGER, + &mpu->mode)) + snd_rawmidi_receive(mpu->substream_input, + (unsigned char *)&val, 1); + + wTmp = readw(mpu->dev->MIDQ + JQS_wHead) + 1; + if (wTmp > readw(mpu->dev->MIDQ + JQS_wSize)) + writew(0, mpu->dev->MIDQ + JQS_wHead); + else + writew(wTmp, mpu->dev->MIDQ + JQS_wHead); + } + spin_unlock_irqrestore(&mpu->input_lock, flags); +} + +static struct snd_rawmidi_ops snd_msndmidi_input = { + .open = snd_msndmidi_input_open, + .close = snd_msndmidi_input_close, + .trigger = snd_msndmidi_input_trigger, +}; + +static void snd_msndmidi_free(struct snd_rawmidi *rmidi) +{ + struct snd_msndmidi *mpu = rmidi->private_data; + kfree(mpu); +} + +int snd_msndmidi_new(struct snd_card *card, int device) +{ + struct snd_msnd *chip = card->private_data; + struct snd_msndmidi *mpu; + struct snd_rawmidi *rmidi; + int err; + + err = snd_rawmidi_new(card, "MSND-MIDI", device, 1, 1, &rmidi); + if (err < 0) + return err; + mpu = kcalloc(1, sizeof(*mpu), GFP_KERNEL); + if (mpu == NULL) { + snd_device_free(card, rmidi); + return -ENOMEM; + } + mpu->dev = chip; + chip->msndmidi_mpu = mpu; + rmidi->private_data = mpu; + rmidi->private_free = snd_msndmidi_free; + spin_lock_init(&mpu->input_lock); + strcpy(rmidi->name, "MSND MIDI"); + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &snd_msndmidi_input); + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; + return 0; +} diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c new file mode 100644 index 0000000..788b1ad --- /dev/null +++ b/sound/isa/msnd/msnd_pinnacle.c @@ -0,0 +1,1787 @@ +/********************************************************************* + * + * Linux multisound pinnacle/fiji driver for ALSA. + * + * 2002/06/30 Karsten Wiese: + * for now this is only used to build a pinnacle / fiji driver. + * the OSS parent of this code is designed to also support + * the multisound classic via the file msnd_classic.c. + * to make it easier for some brave heart to implemt classic + * support in alsa, i left all the MSND_CLASSIC tokens in this file. + * but for now this untested & undone. + * + * + * ripped from linux kernel 2.4.18 by Karsten Wiese. + * + * the following is a copy of the 2.4.18 OSS FREE file-heading comment: + * + * Turtle Beach MultiSound Sound Card Driver for Linux + * msnd_pinnacle.c / msnd_classic.c + * + * -- If MSND_CLASSIC is defined: + * + * -> driver for Turtle Beach Classic/Monterey/Tahiti + * + * -- Else + * + * -> driver for Turtle Beach Pinnacle/Fiji + * + * 12-3-2000 Modified IO port validation Steve Sycamore + * + * Copyright (C) 1998 Andrew Veliath + * + * 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., 675 Mass Ave, Cambridge, MA 02139, USA. + * + ********************************************************************/ + +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/types.h> +#include <linux/delay.h> +#include <linux/ioport.h> +#include <linux/firmware.h> +#include <linux/isa.h> +#include <linux/isapnp.h> +#include <linux/irq.h> +#include <linux/io.h> + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/asound.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/mpu401.h> + +#ifdef MSND_CLASSIC +# ifndef __alpha__ +# define SLOWIO +# endif +#endif +#include "msnd.h" +#ifdef MSND_CLASSIC +# include "msnd_classic.h" +# define LOGNAME "msnd_classic" +#else +# include "msnd_pinnacle.h" +# define LOGNAME "snd_msnd_pinnacle" +#endif + +#ifdef CONFIG_MSNDCLAS_HAVE_BOOT +# define HAVE_DSPCODEH +#endif + +static inline long get_play_delay_jiffies(struct snd_msnd *chip, long size) +{ + long tmp = (size * HZ * chip->play_sample_size) / 8; + return tmp / (chip->play_sample_rate * chip->play_channels); +} + +static inline long get_rec_delay_jiffies(struct snd_msnd *chip, long size) +{ + long tmp = (size * HZ * chip->capture_sample_size) / 8; + return tmp / (chip->capture_sample_rate * chip->capture_channels); +} + +#ifndef HAVE_DSPCODEH +static const u8 *dspini, *permini; +static int sizeof_dspini, sizeof_permini; +#endif + +static int snd_msnd_dsp_full_reset(struct snd_card *card); + +int snd_msnd_send_dsp_cmd_chk(struct snd_msnd *chip, u8 cmd) +{ + if (snd_msnd_send_dsp_cmd(chip, cmd) == 0) + return 0; + snd_msnd_dsp_full_reset(chip->card); + return snd_msnd_send_dsp_cmd(chip, cmd); +} + +static void snd_msnd_play_reset_queue(struct snd_msnd *chip, + unsigned int pcm_periods, + unsigned int pcm_count) +{ + int n; + void *pDAQ = chip->mappedbase + DAPQ_DATA_BUFF; + + chip->last_playbank = -1; + chip->playLimit = pcm_count * (pcm_periods - 1); + chip->playPeriods = pcm_periods; + writew(PCTODSP_OFFSET(0 * DAQDS__size), chip->DAPQ + JQS_wHead); + writew(PCTODSP_OFFSET(0 * DAQDS__size), chip->DAPQ + JQS_wTail); + + chip->play_period_bytes = pcm_count; + + for (n = 0; n < pcm_periods; ++n, pDAQ += DAQDS__size) { + writew(PCTODSP_BASED((u32)(pcm_count * (n % pcm_periods))), + pDAQ + DAQDS_wStart); + writew(0, pDAQ + DAQDS_wSize); + writew(1, pDAQ + DAQDS_wFormat); + writew(chip->play_sample_size, pDAQ + DAQDS_wSampleSize); + writew(chip->play_channels, pDAQ + DAQDS_wChannels); + writew(chip->play_sample_rate, pDAQ + DAQDS_wSampleRate); + writew(HIMT_PLAY_DONE * 0x100 + n, pDAQ + DAQDS_wIntMsg); + writew(n, pDAQ + DAQDS_wFlags); + } +} + +static void snd_msnd_capture_reset_queue(struct snd_msnd *chip, + unsigned int pcm_periods, + unsigned int pcm_count) +{ + int n; + void *pDAQ; + /* unsigned long flags; */ + + /* snd_msnd_init_queue(chip->DARQ, DARQ_DATA_BUFF, DARQ_BUFF_SIZE); */ + + chip->last_recbank = 2; + chip->captureLimit = pcm_count * (pcm_periods - 1); + chip->capturePeriods = pcm_periods; + writew(PCTODSP_OFFSET(0 * DAQDS__size), chip->DARQ + JQS_wHead); + writew(PCTODSP_OFFSET(chip->last_recbank * DAQDS__size), + chip->DARQ + JQS_wTail); + + /* Critical section: bank 1 access. this is how the OSS driver does it: + spin_lock_irqsave(&chip->lock, flags); + outb(HPBLKSEL_1, chip->io + HP_BLKS); + memset_io(chip->mappedbase, 0, DAR_BUFF_SIZE * 3); + outb(HPBLKSEL_0, chip->io + HP_BLKS); + spin_unlock_irqrestore(&chip->lock, flags);*/ + + chip->capturePeriodBytes = pcm_count; + snd_printdd("snd_msnd_capture_reset_queue() %i\n", pcm_count); + + pDAQ = chip->mappedbase + DARQ_DATA_BUFF; + + for (n = 0; n < pcm_periods; ++n, pDAQ += DAQDS__size) { + u32 tmp = pcm_count * (n % pcm_periods); + + writew(PCTODSP_BASED(tmp + 0x3000), pDAQ + DAQDS_wStart); + writew(pcm_count, pDAQ + DAQDS_wSize); + writew(1, pDAQ + DAQDS_wFormat); + writew(chip->capture_sample_size, pDAQ + DAQDS_wSampleSize); + writew(chip->capture_channels, pDAQ + DAQDS_wChannels); + writew(chip->capture_sample_rate, pDAQ + DAQDS_wSampleRate); + writew(HIMT_RECORD_DONE * 0x100 + n, pDAQ + DAQDS_wIntMsg); + writew(n, pDAQ + DAQDS_wFlags); + } +} + +static void dsp_write_flush(struct snd_msnd *chip) +{ + if (!(chip->mode & FMODE_WRITE) || !test_bit(F_WRITING, &chip->flags)) + return; + set_bit(F_WRITEFLUSH, &chip->flags); +/* interruptible_sleep_on_timeout( + &chip->writeflush, + get_play_delay_jiffies(&chip, chip->DAPF.len));*/ + clear_bit(F_WRITEFLUSH, &chip->flags); + if (!signal_pending(current)) + schedule_timeout_interruptible( + get_play_delay_jiffies(chip, chip->play_period_bytes)); + clear_bit(F_WRITING, &chip->flags); +} + +static void dsp_halt(struct snd_msnd *chip, struct file *file) +{ + if ((file ? file->f_mode : chip->mode) & FMODE_READ) { + clear_bit(F_READING, &chip->flags); + snd_msnd_send_dsp_cmd_chk(chip, HDEX_RECORD_STOP); + snd_msnd_disable_irq(chip); + if (file) { + snd_printd(LOGNAME ": Stopping read for %p\n", file); + chip->mode &= ~FMODE_READ; + } + clear_bit(F_AUDIO_READ_INUSE, &chip->flags); + } + if ((file ? file->f_mode : chip->mode) & FMODE_WRITE) { + if (test_bit(F_WRITING, &chip->flags)) { + dsp_write_flush(chip); + snd_msnd_send_dsp_cmd_chk(chip, HDEX_PLAY_STOP); + } + snd_msnd_disable_irq(chip); + if (file) { + snd_printd(LOGNAME ": Stopping write for %p\n", file); + chip->mode &= ~FMODE_WRITE; + } + clear_bit(F_AUDIO_WRITE_INUSE, &chip->flags); + } +} + +#ifdef NO0 +static int dsp_release(struct file *file) +{ + dsp_halt(file); + return 0; +} +#endif + + +static void set_default_audio_parameters(struct snd_msnd *chip) +{ + chip->play_sample_size = DEFSAMPLESIZE; + chip->play_sample_rate = DEFSAMPLERATE; + chip->play_channels = DEFCHANNELS; + chip->capture_sample_size = DEFSAMPLESIZE; + chip->capture_sample_rate = DEFSAMPLERATE; + chip->capture_channels = DEFCHANNELS; +} + + +static int snd_msnd_DARQ(struct snd_msnd *chip, int bank) +{ + int /*size, n,*/ timeout = 3; + u16 wTmp; + /* void *DAQD; */ + + /* Increment the tail and check for queue wrap */ + wTmp = readw(chip->DARQ + JQS_wTail) + PCTODSP_OFFSET(DAQDS__size); + if (wTmp > readw(chip->DARQ + JQS_wSize)) + wTmp = 0; + while (wTmp == readw(chip->DARQ + JQS_wHead) && timeout--) + udelay(1); + + if (chip->capturePeriods == 2) { + void *pDAQ = chip->mappedbase + DARQ_DATA_BUFF + + bank * DAQDS__size + DAQDS_wStart; + unsigned short offset = 0x3000 + chip->capturePeriodBytes; + + if (readw(pDAQ) != PCTODSP_BASED(0x3000)) + offset = 0x3000; + writew(PCTODSP_BASED(offset), pDAQ); + } + + writew(wTmp, chip->DARQ + JQS_wTail); + + /* Get our digital audio queue struct + DAQD = bank * DAQDS__size + chip->mappedbase + DARQ_DATA_BUFF; + + / * Get length of data * / + size = readw(DAQD + DAQDS_wSize); + + / * Read data from the head (unprotected bank 1 access okay + since this is only called inside an interrupt) * / + outb(HPBLKSEL_1, chip->io + HP_BLKS); + if ((n = msnd_fifo_write( + &chip->DARF, + (char *)(chip->base + bank * DAR_BUFF_SIZE), + size, 0)) <= 0) { + outb(HPBLKSEL_0, chip->io + HP_BLKS); + return n; + } + outb(HPBLKSEL_0, chip->io + HP_BLKS); + */ + + return 1; +} + +static int snd_msnd_DAPQ(struct snd_msnd *chip, int start) +{ + u16 DAPQ_tail; + int protect = start, nbanks = 0; + void *DAQD; + static int play_banks_submitted; + /* unsigned long flags; + spin_lock_irqsave(&chip->lock, flags); not necessary */ + + DAPQ_tail = readw(chip->DAPQ + JQS_wTail); + while (DAPQ_tail != readw(chip->DAPQ + JQS_wHead) || start) { + int bank_num = DAPQ_tail / PCTODSP_OFFSET(DAQDS__size); + + if (start) { + start = 0; + play_banks_submitted = 0; + } + + /* Get our digital audio queue struct */ + DAQD = bank_num * DAQDS__size + chip->mappedbase + + DAPQ_DATA_BUFF; + + /* Write size of this bank */ + writew(chip->play_period_bytes, DAQD + DAQDS_wSize); + if (play_banks_submitted < 3) + ++play_banks_submitted; + else if (chip->playPeriods == 2) { + unsigned short offset = chip->play_period_bytes; + + if (readw(DAQD + DAQDS_wStart) != PCTODSP_BASED(0x0)) + offset = 0; + + writew(PCTODSP_BASED(offset), DAQD + DAQDS_wStart); + } + ++nbanks; + + /* Then advance the tail */ + /* + if (protect) + snd_printd("B %X %lX\n", bank_num, xtime.tv_usec); + */ + + DAPQ_tail = (++bank_num % 3) * PCTODSP_OFFSET(DAQDS__size); + writew(DAPQ_tail, chip->DAPQ + JQS_wTail); + /* Tell the DSP to play the bank */ + snd_msnd_send_dsp_cmd(chip, HDEX_PLAY_START); + if (protect) + if (2 == bank_num) + break; + } + /* + if (protect) + snd_printd("%lX\n", xtime.tv_usec); + */ + /* spin_unlock_irqrestore(&chip->lock, flags); not necessary */ + return nbanks; +} + + +static void snd_msnd_eval_dsp_msg(struct snd_msnd *chip, u16 wMessage) +{ + switch (HIBYTE(wMessage)) { + case HIMT_PLAY_DONE: { + #ifdef CONFIG_SND_DEBUG0 + if (chip->banksPlayed < 3) + snd_printd("%08X: HIMT_PLAY_DONE: %i\n", + (unsigned)jiffies, LOBYTE(wMessage)); + #endif + + if (chip->last_playbank == LOBYTE(wMessage)) { + snd_printd("chip.last_playbank == LOBYTE(wMessage)\n"); + break; + } + chip->banksPlayed++; + + if (test_bit(F_WRITING, &chip->flags)) + snd_msnd_DAPQ(chip, 0); + + chip->last_playbank = LOBYTE(wMessage); + chip->playDMAPos += chip->play_period_bytes; + if (chip->playDMAPos > chip->playLimit) + chip->playDMAPos = 0; + snd_pcm_period_elapsed(chip->playback_substream); + + break; + } + case HIMT_RECORD_DONE: + if (chip->last_recbank == LOBYTE(wMessage)) + break; + chip->last_recbank = LOBYTE(wMessage); + chip->captureDMAPos += chip->capturePeriodBytes; + if (chip->captureDMAPos > (chip->captureLimit)) + chip->captureDMAPos = 0; + + if (test_bit(F_READING, &chip->flags)) + snd_msnd_DARQ(chip, chip->last_recbank); + + snd_pcm_period_elapsed(chip->capture_substream); + break; + + case HIMT_DSP: + switch (LOBYTE(wMessage)) { +#ifndef MSND_CLASSIC + case HIDSP_PLAY_UNDER: +#endif + case HIDSP_INT_PLAY_UNDER: + snd_printd(LOGNAME ": Play underflow %i\n", + chip->banksPlayed); + if (chip->banksPlayed > 2) + clear_bit(F_WRITING, &chip->flags); + break; + + case HIDSP_INT_RECORD_OVER: + snd_printd(LOGNAME ": Record overflow\n"); + clear_bit(F_READING, &chip->flags); + break; + + default: + snd_printd(LOGNAME ": DSP message %d 0x%02x\n", + LOBYTE(wMessage), LOBYTE(wMessage)); + break; + } + break; + + case HIMT_MIDI_IN_UCHAR: + if (chip->msndmidi_mpu) + snd_msndmidi_input_read(chip->msndmidi_mpu); + break; + + default: + snd_printd(LOGNAME ": HIMT message %d 0x%02x\n", + HIBYTE(wMessage), HIBYTE(wMessage)); + break; + } +} + +irqreturn_t snd_msnd_interrupt(int irq, void *dev_id) +{ + struct snd_msnd *chip = dev_id; + void *pwDSPQData = chip->mappedbase + DSPQ_DATA_BUFF; + + /* Send ack to DSP */ + /* inb(chip->io + HP_RXL); */ + + /* Evaluate queued DSP messages */ + while (readw(chip->DSPQ + JQS_wTail) != readw(chip->DSPQ + JQS_wHead)) { + u16 wTmp; + + snd_msnd_eval_dsp_msg(chip, + readw(pwDSPQData + 2 * readw(chip->DSPQ + JQS_wHead))); + + wTmp = readw(chip->DSPQ + JQS_wHead) + 1; + if (wTmp > readw(chip->DSPQ + JQS_wSize)) + writew(0, chip->DSPQ + JQS_wHead); + else + writew(wTmp, chip->DSPQ + JQS_wHead); + } + /* Send ack to DSP */ + inb(chip->io + HP_RXL); + return IRQ_HANDLED; +} + + +static int snd_msnd_reset_dsp(long io, unsigned char *info) +{ + int timeout = 100; + + outb(HPDSPRESET_ON, io + HP_DSPR); + msleep(1); +#ifndef MSND_CLASSIC + if (info) + *info = inb(io + HP_INFO); +#endif + outb(HPDSPRESET_OFF, io + HP_DSPR); + msleep(1); + while (timeout-- > 0) { + if (inb(io + HP_CVR) == HP_CVR_DEF) + return 0; + msleep(1); + } + snd_printk(KERN_ERR LOGNAME ": Cannot reset DSP\n"); + + return -EIO; +} + +static int __devinit snd_msnd_probe(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + unsigned char info; +#ifndef MSND_CLASSIC + char *xv, *rev = NULL; + char *pin = "TB Pinnacle", *fiji = "TB Fiji"; + char *pinfiji = "TB Pinnacle/Fiji"; +#endif + + if (!request_region(chip->io, DSP_NUMIO, "probing")) { + snd_printk(KERN_ERR LOGNAME ": I/O port conflict\n"); + return -ENODEV; + } + + if (snd_msnd_reset_dsp(chip->io, &info) < 0) { + release_region(chip->io, DSP_NUMIO); + return -ENODEV; + } + +#ifdef MSND_CLASSIC + strcpy(card->shortname, "Classic/Tahiti/Monterey"); + printk(KERN_INFO LOGNAME ": %s, " +#else + switch (info >> 4) { + case 0xf: + xv = "<= 1.15"; + break; + case 0x1: + xv = "1.18/1.2"; + break; + case 0x2: + xv = "1.3"; + break; + case 0x3: + xv = "1.4"; + break; + default: + xv = "unknown"; + break; + } + + switch (info & 0x7) { + case 0x0: + rev = "I"; + strcpy(card->shortname, pin); + break; + case 0x1: + rev = "F"; + strcpy(card->shortname, pin); + break; + case 0x2: + rev = "G"; + strcpy(card->shortname, pin); + break; + case 0x3: + rev = "H"; + strcpy(card->shortname, pin); + break; + case 0x4: + rev = "E"; + strcpy(card->shortname, fiji); + break; + case 0x5: + rev = "C"; + strcpy(card->shortname, fiji); + break; + case 0x6: + rev = "D"; + strcpy(card->shortname, fiji); + break; + case 0x7: + rev = "A-B (Fiji) or A-E (Pinnacle)"; + strcpy(card->shortname, pinfiji); + break; + } + printk(KERN_INFO LOGNAME ": %s revision %s, Xilinx version %s, " +#endif /* MSND_CLASSIC */ + "I/O 0x%lx-0x%lx, IRQ %d, memory mapped to 0x%lX-0x%lX\n", + card->shortname, +#ifndef MSND_CLASSIC + rev, xv, +#endif + chip->io, chip->io + DSP_NUMIO - 1, + chip->irq, + chip->base, chip->base + 0x7fff); + + strcpy(card->longname, "Turtle Beach Multisound Pinnacle"); + release_region(chip->io, DSP_NUMIO); + return 0; +} + +static int snd_msnd_init_sma(struct snd_msnd *chip) +{ + static int initted; + u16 mastVolLeft, mastVolRight; + unsigned long flags; + +#ifdef MSND_CLASSIC + outb(chip->memid, chip->io + HP_MEMM); +#endif + outb(HPBLKSEL_0, chip->io + HP_BLKS); + /* Motorola 56k shared memory base */ + chip->SMA = chip->mappedbase + SMA_STRUCT_START; + + if (initted) { + mastVolLeft = readw(chip->SMA + SMA_wCurrMastVolLeft); + mastVolRight = readw(chip->SMA + SMA_wCurrMastVolRight); + } else + mastVolLeft = mastVolRight = 0; + memset_io(chip->mappedbase, 0, 0x8000); + + /* Critical section: bank 1 access */ + spin_lock_irqsave(&chip->lock, flags); + outb(HPBLKSEL_1, chip->io + HP_BLKS); + memset_io(chip->mappedbase, 0, 0x8000); + outb(HPBLKSEL_0, chip->io + HP_BLKS); + spin_unlock_irqrestore(&chip->lock, flags); + + /* Digital audio play queue */ + chip->DAPQ = chip->mappedbase + DAPQ_OFFSET; + snd_msnd_init_queue(chip->DAPQ, DAPQ_DATA_BUFF, DAPQ_BUFF_SIZE); + + /* Digital audio record queue */ + chip->DARQ = chip->mappedbase + DARQ_OFFSET; + snd_msnd_init_queue(chip->DARQ, DARQ_DATA_BUFF, DARQ_BUFF_SIZE); + + /* MIDI out queue */ + chip->MODQ = chip->mappedbase + MODQ_OFFSET; + snd_msnd_init_queue(chip->MODQ, MODQ_DATA_BUFF, MODQ_BUFF_SIZE); + + /* MIDI in queue */ + chip->MIDQ = chip->mappedbase + MIDQ_OFFSET; + snd_msnd_init_queue(chip->MIDQ, MIDQ_DATA_BUFF, MIDQ_BUFF_SIZE); + + /* DSP -> host message queue */ + chip->DSPQ = chip->mappedbase + DSPQ_OFFSET; + snd_msnd_init_queue(chip->DSPQ, DSPQ_DATA_BUFF, DSPQ_BUFF_SIZE); + + /* Setup some DSP values */ +#ifndef MSND_CLASSIC + writew(1, chip->SMA + SMA_wCurrPlayFormat); + writew(chip->play_sample_size, chip->SMA + SMA_wCurrPlaySampleSize); + writew(chip->play_channels, chip->SMA + SMA_wCurrPlayChannels); + writew(chip->play_sample_rate, chip->SMA + SMA_wCurrPlaySampleRate); +#endif + writew(chip->play_sample_rate, chip->SMA + SMA_wCalFreqAtoD); + writew(mastVolLeft, chip->SMA + SMA_wCurrMastVolLeft); + writew(mastVolRight, chip->SMA + SMA_wCurrMastVolRight); +#ifndef MSND_CLASSIC + writel(0x00010000, chip->SMA + SMA_dwCurrPlayPitch); + writel(0x00000001, chip->SMA + SMA_dwCurrPlayRate); +#endif + writew(0x303, chip->SMA + SMA_wCurrInputTagBits); + + initted = 1; + + return 0; +} + + +static int __devinit snd_msnd_calibrate_adc(struct snd_msnd *chip, u16 srate) +{ + snd_printd("snd_msnd_calibrate_adc(%i)\n", srate); + writew(srate, chip->SMA + SMA_wCalFreqAtoD); + if (chip->calibrate_signal == 0) + writew(readw(chip->SMA + SMA_wCurrHostStatusFlags) + | 0x0001, chip->SMA + SMA_wCurrHostStatusFlags); + else + writew(readw(chip->SMA + SMA_wCurrHostStatusFlags) + & ~0x0001, chip->SMA + SMA_wCurrHostStatusFlags); + if (snd_msnd_send_word(chip, 0, 0, HDEXAR_CAL_A_TO_D) == 0 && + snd_msnd_send_dsp_cmd_chk(chip, HDEX_AUX_REQ) == 0) { + schedule_timeout_interruptible(msecs_to_jiffies(333)); + return 0; + } + printk(KERN_WARNING LOGNAME ": ADC calibration failed\n"); + return -EIO; +} + +static int upload_dsp_code(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; +#ifndef HAVE_DSPCODEH + const struct firmware *init_fw = NULL, *perm_fw = NULL; +#endif + int err; + + outb(HPBLKSEL_0, chip->io + HP_BLKS); +#ifndef HAVE_DSPCODEH + + err = request_firmware(&init_fw, INITCODEFILE, card->dev); + if (err < 0) { + printk(KERN_ERR LOGNAME ": Error loading " INITCODEFILE); + goto cleanup1; + } + err = request_firmware(&perm_fw, PERMCODEFILE, card->dev); + if (err < 0) { + printk(KERN_ERR LOGNAME ": Error loading " PERMCODEFILE); + goto cleanup; + } + INITCODE = init_fw->data; + INITCODESIZE = init_fw->size; + PERMCODE = perm_fw->data; + PERMCODESIZE = perm_fw->size; +#endif + memcpy_toio(chip->mappedbase, PERMCODE, PERMCODESIZE); + if (snd_msnd_upload_host(chip, INITCODE, INITCODESIZE) < 0) { + printk(KERN_WARNING LOGNAME ": Error uploading to DSP\n"); + err = -ENODEV; + goto cleanup; + } +#ifdef HAVE_DSPsleep + printk(KERN_INFO LOGNAME ": DSP firmware uploaded (resident)\n"); +#else + printk(KERN_INFO LOGNAME ": DSP firmware uploaded\n"); +#endif + err = 0; + +cleanup: +#ifndef HAVE_DSPCODEH + release_firmware(perm_fw); +cleanup1: + release_firmware(init_fw); +#endif + return err; +} + +#ifdef MSND_CLASSIC +static void reset_proteus(struct snd_msnd *chip) +{ + outb(HPPRORESET_ON, chip->io + HP_PROR); + msleep(TIME_PRO_RESET); + outb(HPPRORESET_OFF, chip->io + HP_PROR); + msleep(TIME_PRO_RESET_DONE); +} +#endif + +static int snd_msnd_initialize(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + int err, timeout; + +#ifdef MSND_CLASSIC + outb(HPWAITSTATE_0, chip->io + HP_WAIT); + outb(HPBITMODE_16, chip->io + HP_BITM); + + reset_proteus(chip); +#endif + err = snd_msnd_init_sma(chip); + if (err < 0) { + printk(KERN_WARNING LOGNAME ": Cannot initialize SMA\n"); + return err; + } + + err = snd_msnd_reset_dsp(chip->io, NULL); + if (err < 0) + return err; + + err = upload_dsp_code(card); + if (err < 0) { + printk(KERN_WARNING LOGNAME ": Cannot upload DSP code\n"); + return err; + } + + timeout = 200; + + while (readw(chip->mappedbase)) { + msleep(1); + if (!timeout--) { + snd_printd(LOGNAME ": DSP reset timeout\n"); + return -EIO; + } + } + + snd_msndmix_setup(chip); + return 0; +} + +static int snd_msnd_dsp_full_reset(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + int rv; + + if (test_bit(F_RESETTING, &chip->flags) || ++chip->nresets > 10) + return 0; + + set_bit(F_RESETTING, &chip->flags); + dsp_halt(chip, NULL); /* Unconditionally halt */ + + rv = snd_msnd_initialize(card); + if (rv) + printk(KERN_WARNING LOGNAME ": DSP reset failed\n"); + snd_msndmix_force_recsrc(chip, 0); + clear_bit(F_RESETTING, &chip->flags); + return rv; +} + +static int snd_msnd_dev_free(struct snd_device *device) +{ + snd_printd("snd_msnd_chip_free()\n"); + return 0; +} + +static struct snd_pcm_hardware snd_msnd_playback = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID, + .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = 0x3000, + .period_bytes_min = 0x40, + .period_bytes_max = 0x1800, + .periods_min = 2, + .periods_max = 3, + .fifo_size = 0, +}; + +static struct snd_pcm_hardware snd_msnd_capture = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID, + .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = 0x3000, + .period_bytes_min = 0x40, + .period_bytes_max = 0x1800, + .periods_min = 2, + .periods_max = 3, + .fifo_size = 0, +}; + + +static unsigned int rates[] = { + 8000, 11025, 16000, 22050, 32000, 44100, 48000 +}; + +static struct snd_pcm_hw_constraint_list hw_constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static int snd_msnd_playback_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + set_bit(F_AUDIO_WRITE_INUSE, &chip->flags); + clear_bit(F_WRITING, &chip->flags); + snd_msnd_enable_irq(chip); + + runtime->dma_area = chip->mappedbase; + runtime->dma_bytes = 0x3000; + + chip->playback_substream = substream; + runtime->hw = snd_msnd_playback; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &hw_constraints_rates); + return 0; +} + +static int snd_msnd_playback_close(struct snd_pcm_substream *substream) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + snd_msnd_disable_irq(chip); + clear_bit(F_AUDIO_WRITE_INUSE, &chip->flags); + return 0; +} + + +static int snd_msnd_playback_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int i; + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + void *pDAQ = chip->mappedbase + DAPQ_DATA_BUFF; + + chip->play_sample_size = snd_pcm_format_width(params_format(params)); + chip->play_channels = params_channels(params); + chip->play_sample_rate = params_rate(params); + + for (i = 0; i < 3; ++i, pDAQ += DAQDS__size) { + writew(chip->play_sample_size, pDAQ + DAQDS_wSampleSize); + writew(chip->play_channels, pDAQ + DAQDS_wChannels); + writew(chip->play_sample_rate, pDAQ + DAQDS_wSampleRate); + } + /* dont do this here: + * snd_msnd_calibrate_adc(chip->play_sample_rate); + */ + + return 0; +} + +static int snd_msnd_playback_prepare(struct snd_pcm_substream *substream) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + unsigned int pcm_size = snd_pcm_lib_buffer_bytes(substream); + unsigned int pcm_count = snd_pcm_lib_period_bytes(substream); + unsigned int pcm_periods = pcm_size / pcm_count; + + snd_msnd_play_reset_queue(chip, pcm_periods, pcm_count); + chip->playDMAPos = 0; + return 0; +} + +static int snd_msnd_playback_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + int result = 0; + + if (cmd == SNDRV_PCM_TRIGGER_START) { + snd_printdd("snd_msnd_playback_trigger(START)\n"); + chip->banksPlayed = 0; + set_bit(F_WRITING, &chip->flags); + snd_msnd_DAPQ(chip, 1); + } else if (cmd == SNDRV_PCM_TRIGGER_STOP) { + snd_printdd("snd_msnd_playback_trigger(STop)\n"); + /* interrupt diagnostic, comment this out later */ + clear_bit(F_WRITING, &chip->flags); + snd_msnd_send_dsp_cmd(chip, HDEX_PLAY_STOP); + } else { + snd_printd("snd_msnd_playback_trigger(?????)\n"); + result = -EINVAL; + } + + snd_printdd("snd_msnd_playback_trigger() ENDE\n"); + return result; +} + +static snd_pcm_uframes_t +snd_msnd_playback_pointer(struct snd_pcm_substream *substream) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + return bytes_to_frames(substream->runtime, chip->playDMAPos); +} + + +static struct snd_pcm_ops snd_msnd_playback_ops = { + .open = snd_msnd_playback_open, + .close = snd_msnd_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_msnd_playback_hw_params, + .prepare = snd_msnd_playback_prepare, + .trigger = snd_msnd_playback_trigger, + .pointer = snd_msnd_playback_pointer, +}; + +static int snd_msnd_capture_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + set_bit(F_AUDIO_READ_INUSE, &chip->flags); + snd_msnd_enable_irq(chip); + runtime->dma_area = chip->mappedbase + 0x3000; + runtime->dma_bytes = 0x3000; + memset(runtime->dma_area, 0, runtime->dma_bytes); + chip->capture_substream = substream; + runtime->hw = snd_msnd_capture; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &hw_constraints_rates); + return 0; +} + +static int snd_msnd_capture_close(struct snd_pcm_substream *substream) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + snd_msnd_disable_irq(chip); + clear_bit(F_AUDIO_READ_INUSE, &chip->flags); + return 0; +} + +static int snd_msnd_capture_prepare(struct snd_pcm_substream *substream) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + unsigned int pcm_size = snd_pcm_lib_buffer_bytes(substream); + unsigned int pcm_count = snd_pcm_lib_period_bytes(substream); + unsigned int pcm_periods = pcm_size / pcm_count; + + snd_msnd_capture_reset_queue(chip, pcm_periods, pcm_count); + chip->captureDMAPos = 0; + return 0; +} + +static int snd_msnd_capture_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + if (cmd == SNDRV_PCM_TRIGGER_START) { + chip->last_recbank = -1; + set_bit(F_READING, &chip->flags); + if (snd_msnd_send_dsp_cmd_chk(chip, HDEX_RECORD_START) == 0) + return 0; + + clear_bit(F_READING, &chip->flags); + } else if (cmd == SNDRV_PCM_TRIGGER_STOP) { + clear_bit(F_READING, &chip->flags); + snd_msnd_send_dsp_cmd(chip, HDEX_RECORD_STOP); + return 0; + } + return -EINVAL; +} + + +static snd_pcm_uframes_t +snd_msnd_capture_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + return bytes_to_frames(runtime, chip->captureDMAPos); +} + + +static int snd_msnd_capture_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int i; + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + void *pDAQ = chip->mappedbase + DARQ_DATA_BUFF; + + chip->capture_sample_size = snd_pcm_format_width(params_format(params)); + chip->capture_channels = params_channels(params); + chip->capture_sample_rate = params_rate(params); + + for (i = 0; i < 3; ++i, pDAQ += DAQDS__size) { + writew(chip->capture_sample_size, pDAQ + DAQDS_wSampleSize); + writew(chip->capture_channels, pDAQ + DAQDS_wChannels); + writew(chip->capture_sample_rate, pDAQ + DAQDS_wSampleRate); + } + return 0; +} + + +static struct snd_pcm_ops snd_msnd_capture_ops = { + .open = snd_msnd_capture_open, + .close = snd_msnd_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_msnd_capture_hw_params, + .prepare = snd_msnd_capture_prepare, + .trigger = snd_msnd_capture_trigger, + .pointer = snd_msnd_capture_pointer, +}; + + +int snd_msnd_pcm(struct snd_card *card, int device, struct snd_pcm **rpcm) +{ + struct snd_msnd *chip = card->private_data; + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(card, "MSNDPINNACLE", device, 1, 1, &pcm); + if (err < 0) + return err; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_msnd_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_msnd_capture_ops); + + pcm->private_data = chip; + strcpy(pcm->name, "Hurricane"); + + + if (rpcm) + *rpcm = pcm; + return 0; +} + +/* + * ALSA callback function, called when attempting to open the MIDI device. + */ +static int snd_msnd_mpu401_open(struct snd_mpu401 *mpu) +{ + snd_msnd_enable_irq(mpu->private_data); + snd_msnd_send_dsp_cmd(mpu->private_data, HDEX_MIDI_IN_START); + return 0; +} + +static void snd_msnd_mpu401_close(struct snd_mpu401 *mpu) +{ + snd_msnd_send_dsp_cmd(mpu->private_data, HDEX_MIDI_IN_STOP); + snd_msnd_disable_irq(mpu->private_data); +} + +static long mpu_io[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT; +static int mpu_irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ; + +static int __devinit snd_msnd_attach(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + int err; + static struct snd_device_ops ops = { + .dev_free = snd_msnd_dev_free, + }; + + err = request_irq(chip->irq, snd_msnd_interrupt, 0, card->shortname, + chip); + if (err < 0) { + printk(KERN_ERR LOGNAME ": Couldn't grab IRQ %d\n", chip->irq); + return err; + } + request_region(chip->io, DSP_NUMIO, card->shortname); + + if (!request_mem_region(chip->base, BUFFSIZE, card->shortname)) { + printk(KERN_ERR LOGNAME + ": unable to grab memory region 0x%lx-0x%lx\n", + chip->base, chip->base + BUFFSIZE - 1); + release_region(chip->io, DSP_NUMIO); + free_irq(chip->irq, chip); + return -EBUSY; + } + chip->mappedbase = ioremap(chip->base, 0x8000); + if (!chip->mappedbase) { + printk(KERN_ERR LOGNAME + ": unable to map memory region 0x%lx-0x%lx\n", + chip->base, chip->base + BUFFSIZE - 1); + err = -EIO; + goto err_release_region; + } + chip->mappedbase = ioremap(chip->base, 0x8000); + + err = snd_msnd_dsp_full_reset(card); + if (err < 0) + goto err_release_region; + + /* Register device */ + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) + goto err_release_region; + + err = snd_msnd_pcm(card, 0, NULL); + if (err < 0) { + printk(KERN_ERR LOGNAME ": error creating new PCM device\n"); + goto err_release_region; + } + + err = snd_msndmix_new(card); + if (err < 0) { + printk(KERN_ERR LOGNAME ": error creating new Mixer device\n"); + goto err_release_region; + } + + + if (mpu_io[0] != SNDRV_AUTO_PORT) { + struct snd_mpu401 *mpu; + + err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, + mpu_io[0], + MPU401_MODE_INPUT | + MPU401_MODE_OUTPUT, + mpu_irq[0], IRQF_DISABLED, + &chip->rmidi); + if (err < 0) { + printk(KERN_ERR LOGNAME + ": error creating new Midi device\n"); + goto err_release_region; + } + mpu = chip->rmidi->private_data; + + mpu->open_input = snd_msnd_mpu401_open; + mpu->close_input = snd_msnd_mpu401_close; + mpu->private_data = chip; + } + + disable_irq(chip->irq); + snd_msnd_calibrate_adc(chip, chip->play_sample_rate); + snd_msndmix_force_recsrc(chip, 0); + + err = snd_card_register(card); + if (err < 0) + goto err_release_region; + + return 0; + +err_release_region: + if (chip->mappedbase) + iounmap(chip->mappedbase); + release_mem_region(chip->base, BUFFSIZE); + release_region(chip->io, DSP_NUMIO); + free_irq(chip->irq, chip); + return err; +} + + +static void __devexit snd_msnd_unload(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + + iounmap(chip->mappedbase); + release_mem_region(chip->base, BUFFSIZE); + release_region(chip->io, DSP_NUMIO); + free_irq(chip->irq, chip); + snd_card_free(card); +} + +#ifndef MSND_CLASSIC + +/* Pinnacle/Fiji Logical Device Configuration */ + +static int __devinit snd_msnd_write_cfg(int cfg, int reg, int value) +{ + outb(reg, cfg); + outb(value, cfg + 1); + if (value != inb(cfg + 1)) { + printk(KERN_ERR LOGNAME ": snd_msnd_write_cfg: I/O error\n"); + return -EIO; + } + return 0; +} + +static int __devinit snd_msnd_write_cfg_io0(int cfg, int num, u16 io) +{ + if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_IO0_BASEHI, HIBYTE(io))) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_IO0_BASELO, LOBYTE(io))) + return -EIO; + return 0; +} + +static int __devinit snd_msnd_write_cfg_io1(int cfg, int num, u16 io) +{ + if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_IO1_BASEHI, HIBYTE(io))) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_IO1_BASELO, LOBYTE(io))) + return -EIO; + return 0; +} + +static int __devinit snd_msnd_write_cfg_irq(int cfg, int num, u16 irq) +{ + if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_IRQ_NUMBER, LOBYTE(irq))) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_IRQ_TYPE, IRQTYPE_EDGE)) + return -EIO; + return 0; +} + +static int __devinit snd_msnd_write_cfg_mem(int cfg, int num, int mem) +{ + u16 wmem; + + mem >>= 8; + wmem = (u16)(mem & 0xfff); + if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_MEMBASEHI, HIBYTE(wmem))) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_MEMBASELO, LOBYTE(wmem))) + return -EIO; + if (wmem && snd_msnd_write_cfg(cfg, IREG_MEMCONTROL, + MEMTYPE_HIADDR | MEMTYPE_16BIT)) + return -EIO; + return 0; +} + +static int __devinit snd_msnd_activate_logical(int cfg, int num) +{ + if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_ACTIVATE, LD_ACTIVATE)) + return -EIO; + return 0; +} + +static int __devinit snd_msnd_write_cfg_logical(int cfg, int num, u16 io0, + u16 io1, u16 irq, int mem) +{ + if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (snd_msnd_write_cfg_io0(cfg, num, io0)) + return -EIO; + if (snd_msnd_write_cfg_io1(cfg, num, io1)) + return -EIO; + if (snd_msnd_write_cfg_irq(cfg, num, irq)) + return -EIO; + if (snd_msnd_write_cfg_mem(cfg, num, mem)) + return -EIO; + if (snd_msnd_activate_logical(cfg, num)) + return -EIO; + return 0; +} + +static int __devinit snd_msnd_pinnacle_cfg_reset(int cfg) +{ + int i; + + /* Reset devices if told to */ + printk(KERN_INFO LOGNAME ": Resetting all devices\n"); + for (i = 0; i < 4; ++i) + if (snd_msnd_write_cfg_logical(cfg, i, 0, 0, 0, 0)) + return -EIO; + + return 0; +} +#endif + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ + +module_param_array(index, int, NULL, S_IRUGO); +MODULE_PARM_DESC(index, "Index value for msnd_pinnacle soundcard."); +module_param_array(id, charp, NULL, S_IRUGO); +MODULE_PARM_DESC(id, "ID string for msnd_pinnacle soundcard."); + +static long io[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT; +static int irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ; +static long mem[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT; + +static long cfg[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT; + +#ifndef MSND_CLASSIC +/* Extra Peripheral Configuration (Default: Disable) */ +static long ide_io0[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT; +static long ide_io1[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT; +static int ide_irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ; + +static long joystick_io[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT; +/* If we have the digital daugherboard... */ +static int digital[SNDRV_CARDS] __devinitdata; + +/* Extra Peripheral Configuration */ +static int reset[SNDRV_CARDS] __devinitdata; +#endif + +static int write_ndelay[SNDRV_CARDS] __devinitdata = + { [0 ... (SNDRV_CARDS-1)] = 1 }; + +#ifdef CONFIG_PNP +static int isapnp[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_ENABLE_PNP; +#endif + +#ifdef MODULE +MODULE_AUTHOR("Karsten Wiese <annabellesgarden(a)yahoo.de>"); +MODULE_DESCRIPTION("Turtle Beach " LONGNAME " Linux Driver"); +MODULE_LICENSE("GPL"); +#ifndef HAVE_DSPCODEH +MODULE_FIRMWARE(INITCODEFILE); +MODULE_FIRMWARE(PERMCODEFILE); +#endif + +static int calibrate_signal __devinitdata; + +module_param_array(io, long, NULL, S_IRUGO); +MODULE_PARM_DESC(io, "IO port #"); +module_param_array(irq, int, NULL, S_IRUGO); +module_param_array(mem, long, NULL, S_IRUGO); +module_param_array(write_ndelay, int, NULL, S_IRUGO); +module_param(calibrate_signal, int, S_IRUGO); +#ifndef MSND_CLASSIC +module_param_array(digital, int, NULL, S_IRUGO); +module_param_array(cfg, long, NULL, S_IRUGO); +module_param_array(reset, int, 0, S_IRUGO); +module_param_array(mpu_io, long, NULL, S_IRUGO); +module_param_array(mpu_irq, int, NULL, S_IRUGO); +module_param_array(ide_io0, long, NULL, S_IRUGO); +module_param_array(ide_io1, long, NULL, S_IRUGO); +module_param_array(ide_irq, int, NULL, S_IRUGO); +module_param_array(joystick_io, long, NULL, S_IRUGO); +#ifdef CONFIG_PNP +module_param_array(isapnp, bool, NULL, 0444); +MODULE_PARM_DESC(isapnp, "ISA PnP detection for specified soundcard."); +#endif +#endif + +#else /* not a module */ + +static int +calibrate_signal __devinitdata; +#endif /* MODULE */ + + +static int __devinit snd_msnd_isa_match(struct device *pdev, unsigned int i) +{ + if (io[i] == SNDRV_AUTO_PORT) + return 0; + + if (irq[i] == SNDRV_AUTO_PORT || mem[i] == SNDRV_AUTO_PORT) { + printk(KERN_WARNING LOGNAME ": io, irq and mem must be set\n"); + return 0; + } + +#ifdef MSND_CLASSIC + if (!(io[i] == 0x290 || + io[i] == 0x260 || + io[i] == 0x250 || + io[i] == 0x240 || + io[i] == 0x230 || + io[i] == 0x220 || + io[i] == 0x210 || + io[i] == 0x3e0)) { + printk(KERN_ERR LOGNAME ": \"io\" - DSP I/O base must be set " + " to 0x210, 0x220, 0x230, 0x240, 0x250, 0x260, 0x290, " + "or 0x3E0\n"); + return 0; + } +#else + if (io[i] < 0x100 || io[i] > 0x3e0 || (io[i] % 0x10) != 0) { + printk(KERN_ERR LOGNAME + ": \"io\" - DSP I/O base must within the range 0x100 " + "to 0x3E0 and must be evenly divisible by 0x10\n"); + return 0; + } +#endif /* MSND_CLASSIC */ + + if (!(irq[i] == 5 || + irq[i] == 7 || + irq[i] == 9 || + irq[i] == 10 || + irq[i] == 11 || + irq[i] == 12)) { + printk(KERN_ERR LOGNAME + ": \"irq\" - must be set to 5, 7, 9, 10, 11 or 12\n"); + return 0; + } + + if (!(mem[i] == 0xb0000 || + mem[i] == 0xc8000 || + mem[i] == 0xd0000 || + mem[i] == 0xd8000 || + mem[i] == 0xe0000 || + mem[i] == 0xe8000)) { + printk(KERN_ERR LOGNAME ": \"mem\" - must be set to " + "0xb0000, 0xc8000, 0xd0000, 0xd8000, 0xe0000 or " + "0xe8000\n"); + return 0; + } + +#ifndef MSND_CLASSIC + if (cfg[i] == SNDRV_AUTO_PORT) { + printk(KERN_INFO LOGNAME ": Assuming PnP mode\n"); + } else if (cfg[i] != 0x250 && cfg[i] != 0x260 && cfg[i] != 0x270) { + printk(KERN_INFO LOGNAME + ": Config port must be 0x250, 0x260 or 0x270 " + "(or unspecified for PnP mode)\n"); + return 0; + } +#endif /* MSND_CLASSIC */ + + return 1; +} + +static int __devinit snd_msnd_isa_probe(struct device *pdev, unsigned int idx) +{ + int err; + struct snd_card *card; + struct snd_msnd *chip; + + if (isapnp[idx] || cfg[idx] == SNDRV_AUTO_PORT) { + printk(KERN_INFO LOGNAME ": Assuming PnP mode\n"); + return -ENODEV; + } + + card = snd_card_new(index[idx], id[idx], THIS_MODULE, + sizeof(struct snd_msnd)); + if (!card) + return -ENOMEM; + + snd_card_set_dev(card, pdev); + chip = card->private_data; + chip->card = card; + +#ifdef MSND_CLASSIC + switch (irq[idx]) { + case 5: + chip->irqid = HPIRQ_5; break; + case 7: + chip->irqid = HPIRQ_7; break; + case 9: + chip->irqid = HPIRQ_9; break; + case 10: + chip->irqid = HPIRQ_10; break; + case 11: + chip->irqid = HPIRQ_11; break; + case 12: + chip->irqid = HPIRQ_12; break; + } + + switch (mem[idx]) { + case 0xb0000: + chip->memid = HPMEM_B000; break; + case 0xc8000: + chip->memid = HPMEM_C800; break; + case 0xd0000: + chip->memid = HPMEM_D000; break; + case 0xd8000: + chip->memid = HPMEM_D800; break; + case 0xe0000: + chip->memid = HPMEM_E000; break; + case 0xe8000: + chip->memid = HPMEM_E800; break; + } +#else + printk(KERN_INFO LOGNAME ": Non-PnP mode: configuring at port 0x%lx\n", + cfg[idx]); + + if (!request_region(cfg[idx], 2, "Pinnacle/Fiji Config")) { + printk(KERN_ERR LOGNAME ": Config port 0x%lx conflict\n", + cfg[idx]); + snd_card_free(card); + return -EIO; + } + if (reset[idx]) + if (snd_msnd_pinnacle_cfg_reset(cfg[idx])) { + err = -EIO; + goto cfg_error; + } + + /* DSP */ + err = snd_msnd_write_cfg_logical(cfg[idx], 0, + io[idx], 0, + irq[idx], mem[idx]); + + if (err) + goto cfg_error; + + /* The following are Pinnacle specific */ + + /* MPU */ + if (mpu_io[idx] != SNDRV_AUTO_PORT + && mpu_irq[idx] != SNDRV_AUTO_IRQ) { + printk(KERN_INFO LOGNAME + ": Configuring MPU to I/O 0x%lx IRQ %d\n", + mpu_io[idx], mpu_irq[idx]); + err = snd_msnd_write_cfg_logical(cfg[idx], 1, + mpu_io[idx], 0, + mpu_irq[idx], 0); + + if (err) + goto cfg_error; + } + + /* IDE */ + if (ide_io0[idx] != SNDRV_AUTO_PORT + && ide_io1[idx] != SNDRV_AUTO_PORT + && ide_irq[idx] != SNDRV_AUTO_IRQ) { + printk(KERN_INFO LOGNAME + ": Configuring IDE to I/O 0x%lx, 0x%lx IRQ %d\n", + ide_io0[idx], ide_io1[idx], ide_irq[idx]); + err = snd_msnd_write_cfg_logical(cfg[idx], 2, + ide_io0[idx], ide_io1[idx], + ide_irq[idx], 0); + + if (err) + goto cfg_error; + } + + /* Joystick */ + if (joystick_io[idx] != SNDRV_AUTO_PORT) { + printk(KERN_INFO LOGNAME + ": Configuring joystick to I/O 0x%lx\n", + joystick_io[idx]); + err = snd_msnd_write_cfg_logical(cfg[idx], 3, + joystick_io[idx], 0, + 0, 0); + + if (err) + goto cfg_error; + } + release_region(cfg[idx], 2); + +#endif /* MSND_CLASSIC */ + + set_default_audio_parameters(chip); +#ifdef MSND_CLASSIC + chip->type = msndClassic; +#else + chip->type = msndPinnacle; +#endif + chip->io = io[idx]; + chip->irq = irq[idx]; + chip->base = mem[idx]; + + chip->calibrate_signal = calibrate_signal ? 1 : 0; + chip->recsrc = 0; + chip->dspq_data_buff = DSPQ_DATA_BUFF; + chip->dspq_buff_size = DSPQ_BUFF_SIZE; + if (write_ndelay[idx]) + clear_bit(F_DISABLE_WRITE_NDELAY, &chip->flags); + else + set_bit(F_DISABLE_WRITE_NDELAY, &chip->flags); +#ifndef MSND_CLASSIC + if (digital[idx]) + set_bit(F_HAVEDIGITAL, &chip->flags); +#endif + spin_lock_init(&chip->lock); + err = snd_msnd_probe(card); + if (err < 0) { + printk(KERN_ERR LOGNAME ": Probe failed\n"); + snd_card_free(card); + return err; + } + + err = snd_msnd_attach(card); + if (err < 0) { + printk(KERN_ERR LOGNAME ": Attach failed\n"); + snd_card_free(card); + return err; + } + dev_set_drvdata(pdev, card); + + return 0; + +#ifndef MSND_CLASSIC +cfg_error: + release_region(cfg[idx], 2); + snd_card_free(card); + return err; +#endif +} + +static int __devexit snd_msnd_isa_remove(struct device *pdev, unsigned int dev) +{ + snd_msnd_unload(dev_get_drvdata(pdev)); + dev_set_drvdata(pdev, NULL); + return 0; +} + +#define DEV_NAME "msnd-pinnacle" + +static struct isa_driver snd_msnd_driver = { + .match = snd_msnd_isa_match, + .probe = snd_msnd_isa_probe, + .remove = __devexit_p(snd_msnd_isa_remove), + /* FIXME: suspend, resume */ + .driver = { + .name = DEV_NAME + }, +}; + +#ifdef CONFIG_PNP +static int __devinit snd_msnd_pnp_detect(struct pnp_card_link *pcard, + const struct pnp_card_device_id *pid) +{ + static int idx; + struct pnp_dev *pnp_dev; + struct pnp_dev *mpu_dev; + struct snd_card *card; + struct snd_msnd *chip; + int ret; + + for ( ; idx < SNDRV_CARDS; idx++) { + if (isapnp[idx]) + break; + } + if (idx >= SNDRV_CARDS) + return -ENODEV; + + /* + * Check that we still have room for another sound card ... + */ + pnp_dev = pnp_request_card_device(pcard, pid->devs[0].id, NULL); + if (!pnp_dev) + return -ENODEV; + + mpu_dev = pnp_request_card_device(pcard, pid->devs[1].id, NULL); + if (!mpu_dev) + return -ENODEV; + + if (!pnp_is_active(pnp_dev) && pnp_activate_dev(pnp_dev) < 0) { + printk(KERN_INFO "msnd_pinnacle: device is inactive\n"); + return -EBUSY; + } + + if (!pnp_is_active(mpu_dev) && pnp_activate_dev(mpu_dev) < 0) { + printk(KERN_INFO "msnd_pinnacle: MPU device is inactive\n"); + return -EBUSY; + } + + /* + * Create a new ALSA sound card entry, in anticipation + * of detecting our hardware ... + */ + card = snd_card_new(index[idx], id[idx], THIS_MODULE, + sizeof(struct snd_msnd)); + if (!card) + return -ENOMEM; + + chip = card->private_data; + chip->card = card; + snd_card_set_dev(card, &pcard->card->dev); + + /* + * Read the correct parameters off the ISA PnP bus ... + */ + io[idx] = pnp_port_start(pnp_dev, 0); + irq[idx] = pnp_irq(pnp_dev, 0); + mem[idx] = pnp_mem_start(pnp_dev, 0); + mpu_io[idx] = pnp_port_start(mpu_dev, 0); + mpu_irq[idx] = pnp_irq(mpu_dev, 0); + + set_default_audio_parameters(chip); +#ifdef MSND_CLASSIC + chip->type = msndClassic; +#else + chip->type = msndPinnacle; +#endif + chip->io = io[idx]; + chip->irq = irq[idx]; + chip->base = mem[idx]; + + chip->calibrate_signal = calibrate_signal ? 1 : 0; + chip->recsrc = 0; + chip->dspq_data_buff = DSPQ_DATA_BUFF; + chip->dspq_buff_size = DSPQ_BUFF_SIZE; + if (write_ndelay[idx]) + clear_bit(F_DISABLE_WRITE_NDELAY, &chip->flags); + else + set_bit(F_DISABLE_WRITE_NDELAY, &chip->flags); +#ifndef MSND_CLASSIC + if (digital[idx]) + set_bit(F_HAVEDIGITAL, &chip->flags); +#endif + spin_lock_init(&chip->lock); + ret = snd_msnd_probe(card); + if (ret < 0) { + printk(KERN_ERR LOGNAME ": Probe failed\n"); + goto _release_card; + } + + ret = snd_msnd_attach(card); + if (ret < 0) { + printk(KERN_ERR LOGNAME ": Attach failed\n"); + goto _release_card; + } + + pnp_set_card_drvdata(pcard, card); + ++idx; + return 0; + +_release_card: + snd_card_free(card); + return ret; +} + +static void __devexit snd_msnd_pnp_remove(struct pnp_card_link *pcard) +{ + snd_msnd_unload(pnp_get_card_drvdata(pcard)); + pnp_set_card_drvdata(pcard, NULL); +} + +static int isa_registered; +static int pnp_registered; + +static struct pnp_card_device_id msnd_pnpids[] = { + /* Pinnacle PnP */ + { .id = "BVJ0440", .devs = { { "TBS0000" }, { "TBS0001" } } }, + { .id = "" } /* end */ +}; + +MODULE_DEVICE_TABLE(pnp_card, msnd_pnpids); + +static struct pnp_card_driver msnd_pnpc_driver = { + .flags = PNP_DRIVER_RES_DO_NOT_CHANGE, + .name = "msnd_pinnacle", + .id_table = msnd_pnpids, + .probe = snd_msnd_pnp_detect, + .remove = __devexit_p(snd_msnd_pnp_remove), +}; +#endif /* CONFIG_PNP */ + +static int __init snd_msnd_init(void) +{ + int err; + + err = isa_register_driver(&snd_msnd_driver, SNDRV_CARDS); +#ifdef CONFIG_PNP + if (!err) + isa_registered = 1; + + err = pnp_register_card_driver(&msnd_pnpc_driver); + if (!err) + pnp_registered = 1; + + if (isa_registered) + err = 0; +#endif + return err; +} + +static void __exit snd_msnd_exit(void) +{ +#ifdef CONFIG_PNP + if (pnp_registered) + pnp_unregister_card_driver(&msnd_pnpc_driver); + if (isa_registered) +#endif + isa_unregister_driver(&snd_msnd_driver); +} + +module_init(snd_msnd_init); +module_exit(snd_msnd_exit); + diff --git a/sound/isa/msnd/msnd_pinnacle.h b/sound/isa/msnd/msnd_pinnacle.h new file mode 100644 index 0000000..d1ddbb8 --- /dev/null +++ b/sound/isa/msnd/msnd_pinnacle.h @@ -0,0 +1,222 @@ +/********************************************************************* + * + * msnd_pinnacle.h + * + * Turtle Beach MultiSound Sound Card Driver for Linux + * + * Some parts of this header file were derived from the Turtle Beach + * MultiSound Driver Development Kit. + * + * Copyright (C) 1998 Andrew Veliath + * Copyright (C) 1993 Turtle Beach Systems, Inc. + * + * 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., 675 Mass Ave, Cambridge, MA 02139, USA. + * + ********************************************************************/ +#ifndef __MSND_PINNACLE_H +#define __MSND_PINNACLE_H + +#define DSP_NUMIO 0x08 + +#define IREG_LOGDEVICE 0x07 +#define IREG_ACTIVATE 0x30 +#define LD_ACTIVATE 0x01 +#define LD_DISACTIVATE 0x00 +#define IREG_EECONTROL 0x3F +#define IREG_MEMBASEHI 0x40 +#define IREG_MEMBASELO 0x41 +#define IREG_MEMCONTROL 0x42 +#define IREG_MEMRANGEHI 0x43 +#define IREG_MEMRANGELO 0x44 +#define MEMTYPE_8BIT 0x00 +#define MEMTYPE_16BIT 0x02 +#define MEMTYPE_RANGE 0x00 +#define MEMTYPE_HIADDR 0x01 +#define IREG_IO0_BASEHI 0x60 +#define IREG_IO0_BASELO 0x61 +#define IREG_IO1_BASEHI 0x62 +#define IREG_IO1_BASELO 0x63 +#define IREG_IRQ_NUMBER 0x70 +#define IREG_IRQ_TYPE 0x71 +#define IRQTYPE_HIGH 0x02 +#define IRQTYPE_LOW 0x00 +#define IRQTYPE_LEVEL 0x01 +#define IRQTYPE_EDGE 0x00 + +#define HP_DSPR 0x04 +#define HP_BLKS 0x04 + +#define HPDSPRESET_OFF 2 +#define HPDSPRESET_ON 0 + +#define HPBLKSEL_0 2 +#define HPBLKSEL_1 3 + +#define HIMT_DAT_OFF 0x03 + +#define HIDSP_PLAY_UNDER 0x00 +#define HIDSP_INT_PLAY_UNDER 0x01 +#define HIDSP_SSI_TX_UNDER 0x02 +#define HIDSP_RECQ_OVERFLOW 0x08 +#define HIDSP_INT_RECORD_OVER 0x09 +#define HIDSP_SSI_RX_OVERFLOW 0x0a + +#define HIDSP_MIDI_IN_OVER 0x10 + +#define HIDSP_MIDI_FRAME_ERR 0x11 +#define HIDSP_MIDI_PARITY_ERR 0x12 +#define HIDSP_MIDI_OVERRUN_ERR 0x13 + +#define HIDSP_INPUT_CLIPPING 0x20 +#define HIDSP_MIX_CLIPPING 0x30 +#define HIDSP_DAT_IN_OFF 0x21 + +#define HDEXAR_SET_ANA_IN 0 +#define HDEXAR_CLEAR_PEAKS 1 +#define HDEXAR_IN_SET_POTS 2 +#define HDEXAR_AUX_SET_POTS 3 +#define HDEXAR_CAL_A_TO_D 4 +#define HDEXAR_RD_EXT_DSP_BITS 5 + +#define HDEXAR_SET_SYNTH_IN 4 +#define HDEXAR_READ_DAT_IN 5 +#define HDEXAR_MIC_SET_POTS 6 +#define HDEXAR_SET_DAT_IN 7 + +#define HDEXAR_SET_SYNTH_48 8 +#define HDEXAR_SET_SYNTH_44 9 + +#define TIME_PRO_RESET_DONE 0x028A +#define TIME_PRO_SYSEX 0x001E +#define TIME_PRO_RESET 0x0032 + +#define DAR_BUFF_SIZE 0x1000 + +#define MIDQ_BUFF_SIZE 0x800 +#define DSPQ_BUFF_SIZE 0x5A0 + +#define DSPQ_DATA_BUFF 0x7860 + +#define MOP_WAVEHDR 0 +#define MOP_EXTOUT 1 +#define MOP_HWINIT 0xfe +#define MOP_NONE 0xff +#define MOP_MAX 1 + +#define MIP_EXTIN 0 +#define MIP_WAVEHDR 1 +#define MIP_HWINIT 0xfe +#define MIP_MAX 1 + +/* Pinnacle/Fiji SMA Common Data */ +#define SMA_wCurrPlayBytes 0x0000 +#define SMA_wCurrRecordBytes 0x0002 +#define SMA_wCurrPlayVolLeft 0x0004 +#define SMA_wCurrPlayVolRight 0x0006 +#define SMA_wCurrInVolLeft 0x0008 +#define SMA_wCurrInVolRight 0x000a +#define SMA_wCurrMHdrVolLeft 0x000c +#define SMA_wCurrMHdrVolRight 0x000e +#define SMA_dwCurrPlayPitch 0x0010 +#define SMA_dwCurrPlayRate 0x0014 +#define SMA_wCurrMIDIIOPatch 0x0018 +#define SMA_wCurrPlayFormat 0x001a +#define SMA_wCurrPlaySampleSize 0x001c +#define SMA_wCurrPlayChannels 0x001e +#define SMA_wCurrPlaySampleRate 0x0020 +#define SMA_wCurrRecordFormat 0x0022 +#define SMA_wCurrRecordSampleSize 0x0024 +#define SMA_wCurrRecordChannels 0x0026 +#define SMA_wCurrRecordSampleRate 0x0028 +#define SMA_wCurrDSPStatusFlags 0x002a +#define SMA_wCurrHostStatusFlags 0x002c +#define SMA_wCurrInputTagBits 0x002e +#define SMA_wCurrLeftPeak 0x0030 +#define SMA_wCurrRightPeak 0x0032 +#define SMA_bMicPotPosLeft 0x0034 +#define SMA_bMicPotPosRight 0x0035 +#define SMA_bMicPotMaxLeft 0x0036 +#define SMA_bMicPotMaxRight 0x0037 +#define SMA_bInPotPosLeft 0x0038 +#define SMA_bInPotPosRight 0x0039 +#define SMA_bAuxPotPosLeft 0x003a +#define SMA_bAuxPotPosRight 0x003b +#define SMA_bInPotMaxLeft 0x003c +#define SMA_bInPotMaxRight 0x003d +#define SMA_bAuxPotMaxLeft 0x003e +#define SMA_bAuxPotMaxRight 0x003f +#define SMA_bInPotMaxMethod 0x0040 +#define SMA_bAuxPotMaxMethod 0x0041 +#define SMA_wCurrMastVolLeft 0x0042 +#define SMA_wCurrMastVolRight 0x0044 +#define SMA_wCalFreqAtoD 0x0046 +#define SMA_wCurrAuxVolLeft 0x0048 +#define SMA_wCurrAuxVolRight 0x004a +#define SMA_wCurrPlay1VolLeft 0x004c +#define SMA_wCurrPlay1VolRight 0x004e +#define SMA_wCurrPlay2VolLeft 0x0050 +#define SMA_wCurrPlay2VolRight 0x0052 +#define SMA_wCurrPlay3VolLeft 0x0054 +#define SMA_wCurrPlay3VolRight 0x0056 +#define SMA_wCurrPlay4VolLeft 0x0058 +#define SMA_wCurrPlay4VolRight 0x005a +#define SMA_wCurrPlay1PeakLeft 0x005c +#define SMA_wCurrPlay1PeakRight 0x005e +#define SMA_wCurrPlay2PeakLeft 0x0060 +#define SMA_wCurrPlay2PeakRight 0x0062 +#define SMA_wCurrPlay3PeakLeft 0x0064 +#define SMA_wCurrPlay3PeakRight 0x0066 +#define SMA_wCurrPlay4PeakLeft 0x0068 +#define SMA_wCurrPlay4PeakRight 0x006a +#define SMA_wCurrPlayPeakLeft 0x006c +#define SMA_wCurrPlayPeakRight 0x006e +#define SMA_wCurrDATSR 0x0070 +#define SMA_wCurrDATRXCHNL 0x0072 +#define SMA_wCurrDATTXCHNL 0x0074 +#define SMA_wCurrDATRXRate 0x0076 +#define SMA_dwDSPPlayCount 0x0078 +#define SMA__size 0x007c + +#ifdef HAVE_DSPCODEH +# include "pndsperm.c" +# include "pndspini.c" +# define PERMCODE pndsperm +# define INITCODE pndspini +# define PERMCODESIZE sizeof(pndsperm) +# define INITCODESIZE sizeof(pndspini) +#else +# ifndef CONFIG_MSNDPIN_INIT_FILE +# define CONFIG_MSNDPIN_INIT_FILE "turtlebeach/pndspini.bin" +# endif +# ifndef CONFIG_MSNDPIN_PERM_FILE +# define CONFIG_MSNDPIN_PERM_FILE "turtlebeach/pndsperm.bin" +# endif +# define PERMCODEFILE CONFIG_MSNDPIN_PERM_FILE +# define INITCODEFILE CONFIG_MSNDPIN_INIT_FILE +# define PERMCODE dspini +# define INITCODE permini +# define PERMCODESIZE sizeof_dspini +# define INITCODESIZE sizeof_permini +#endif +#define LONGNAME "MultiSound (Pinnacle/Fiji)" + +int snd_msnd_send_dsp_cmd_chk(struct snd_msnd *chip, register u8 cmd); + +/* functions from external mixer file */ +void snd_msndmix_setup(struct snd_msnd *chip); +unsigned long snd_msndmix_force_recsrc(struct snd_msnd *chip, int recsrc); +int __devinit snd_msndmix_new(struct snd_card *card); + +#endif /* __MSND_PINNACLE_H */ diff --git a/sound/isa/msnd/msnd_pinnacle_mixer.c b/sound/isa/msnd/msnd_pinnacle_mixer.c new file mode 100644 index 0000000..3f26aac --- /dev/null +++ b/sound/isa/msnd/msnd_pinnacle_mixer.c @@ -0,0 +1,345 @@ +/*************************************************************************** + msnd_pinnacle_mixer.c - description + ------------------- + begin : Fre Jun 7 2002 + copyright : (C) 2002 by karsten wiese + email : annabellesgarden(a)yahoo.de + ***************************************************************************/ + +/*************************************************************************** + * * + * This program is free software; you can redistribute it and/or modify * + * it under the terms of the GNU General Public License as published by * + * the Free Software Foundation; either version 2 of the License, or * + * (at your option) any later version. * + * * + ***************************************************************************/ + +#include <linux/io.h> + +#include <sound/core.h> +#include <sound/control.h> +#include "msnd.h" +#include "msnd_pinnacle.h" + + +#define MSND_MIXER_VOLUME 0 +#define MSND_MIXER_PCM 1 +#define MSND_MIXER_AUX 2 /* Input source 1 (aux1) */ +#define MSND_MIXER_IMIX 3 /* Recording monitor */ +#define MSND_MIXER_SYNTH 4 +#define MSND_MIXER_SPEAKER 5 +#define MSND_MIXER_LINE 6 +#define MSND_MIXER_MIC 7 +#define MSND_MIXER_RECLEV 11 /* Recording level */ +#define MSND_MIXER_IGAIN 12 /* Input gain */ +#define MSND_MIXER_OGAIN 13 /* Output gain */ +#define MSND_MIXER_DIGITAL 17 /* Digital (input) 1 */ + +/* Device mask bits */ + +#define MSND_MASK_VOLUME (1 << MSND_MIXER_VOLUME) +#define MSND_MASK_SYNTH (1 << MSND_MIXER_SYNTH) +#define MSND_MASK_PCM (1 << MSND_MIXER_PCM) +#define MSND_MASK_SPEAKER (1 << MSND_MIXER_SPEAKER) +#define MSND_MASK_LINE (1 << MSND_MIXER_LINE) +#define MSND_MASK_MIC (1 << MSND_MIXER_MIC) +#define MSND_MASK_IMIX (1 << MSND_MIXER_IMIX) +#define MSND_MASK_RECLEV (1 << MSND_MIXER_RECLEV) +#define MSND_MASK_IGAIN (1 << MSND_MIXER_IGAIN) +#define MSND_MASK_OGAIN (1 << MSND_MIXER_OGAIN) +#define MSND_MASK_AUX (1 << MSND_MIXER_AUX) +#define MSND_MASK_DIGITAL (1 << MSND_MIXER_DIGITAL) + +static int snd_msndmix_info_mux(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[3] = { + "Analog", "MASS", "SPDIF", + }; + struct snd_msnd *chip = snd_kcontrol_chip(kcontrol); + unsigned items = test_bit(F_HAVEDIGITAL, &chip->flags) ? 3 : 2; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = items; + if (uinfo->value.enumerated.item >= items) + uinfo->value.enumerated.item = items - 1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + return 0; +} + +static int snd_msndmix_get_mux(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_msnd *chip = snd_kcontrol_chip(kcontrol); + /* MSND_MASK_IMIX is the default */ + ucontrol->value.enumerated.item[0] = 0; + + if (chip->recsrc & MSND_MASK_SYNTH) { + ucontrol->value.enumerated.item[0] = 1; + } else if ((chip->recsrc & MSND_MASK_DIGITAL) && + test_bit(F_HAVEDIGITAL, &chip->flags)) { + ucontrol->value.enumerated.item[0] = 2; + } + + + return 0; +} + +static int snd_msndmix_set_mux(struct snd_msnd *chip, int val) +{ + unsigned newrecsrc; + int change; + unsigned char msndbyte; + + switch (val) { + case 0: + newrecsrc = MSND_MASK_IMIX; + msndbyte = HDEXAR_SET_ANA_IN; + break; + case 1: + newrecsrc = MSND_MASK_SYNTH; + msndbyte = HDEXAR_SET_SYNTH_IN; + break; + case 2: + newrecsrc = MSND_MASK_DIGITAL; + msndbyte = HDEXAR_SET_DAT_IN; + break; + default: + return -EINVAL; + } + change = newrecsrc != chip->recsrc; + if (change) { + change = 0; + if (!snd_msnd_send_word(chip, 0, 0, msndbyte)) + if (!snd_msnd_send_dsp_cmd_chk(chip, HDEX_AUX_REQ)) { + chip->recsrc = newrecsrc; + change = 1; + } + } + return change; +} + + +static int snd_msndmix_put_mux(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_msnd *msnd = snd_kcontrol_chip(kcontrol); + return snd_msndmix_set_mux(msnd, ucontrol->value.enumerated.item[0]); +} + + +static int snd_msndmix_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 100; + return 0; +} + +static int snd_msndmix_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_msnd *msnd = snd_kcontrol_chip(kcontrol); + int addr = kcontrol->private_value; + unsigned long flags; + + spin_lock_irqsave(&msnd->mixer_lock, flags); + ucontrol->value.integer.value[0] = msnd->left_levels[addr] * 100; + ucontrol->value.integer.value[0] /= 0xFFFF; + ucontrol->value.integer.value[1] = msnd->right_levels[addr] * 100; + ucontrol->value.integer.value[1] /= 0xFFFF; + spin_unlock_irqrestore(&msnd->mixer_lock, flags); + return 0; +} + + +#define update_volm(a, b) \ + do { \ + writew((dev->left_levels[a] >> 1) * \ + readw(dev->SMA + SMA_wCurrMastVolLeft) / 0xffff, \ + dev->SMA + SMA_##b##Left); \ + writew((dev->right_levels[a] >> 1) * \ + readw(dev->SMA + SMA_wCurrMastVolRight) / 0xffff, \ + dev->SMA + SMA_##b##Right); \ + } while (0); + +#define update_potm(d, s, ar) \ + do { \ + writeb((dev->left_levels[d] >> 8) * \ + readw(dev->SMA + SMA_wCurrMastVolLeft) / 0xffff, \ + dev->SMA + SMA_##s##Left); \ + writeb((dev->right_levels[d] >> 8) * \ + readw(dev->SMA + SMA_wCurrMastVolRight) / 0xffff, \ + dev->SMA + SMA_##s##Right); \ + if (snd_msnd_send_word(dev, 0, 0, ar) == 0) \ + snd_msnd_send_dsp_cmd_chk(dev, HDEX_AUX_REQ); \ + } while (0); + +#define update_pot(d, s, ar) \ + do { \ + writeb(dev->left_levels[d] >> 8, \ + dev->SMA + SMA_##s##Left); \ + writeb(dev->right_levels[d] >> 8, \ + dev->SMA + SMA_##s##Right); \ + if (snd_msnd_send_word(dev, 0, 0, ar) == 0) \ + snd_msnd_send_dsp_cmd_chk(dev, HDEX_AUX_REQ); \ + } while (0); + + +static int snd_msndmix_set(struct snd_msnd *dev, int d, int left, int right) +{ + int bLeft, bRight; + int wLeft, wRight; + int updatemaster = 0; + + if (d >= LEVEL_ENTRIES) + return -EINVAL; + + bLeft = left * 0xff / 100; + wLeft = left * 0xffff / 100; + + bRight = right * 0xff / 100; + wRight = right * 0xffff / 100; + + dev->left_levels[d] = wLeft; + dev->right_levels[d] = wRight; + + switch (d) { + /* master volume unscaled controls */ + case MSND_MIXER_LINE: /* line pot control */ + /* scaled by IMIX in digital mix */ + writeb(bLeft, dev->SMA + SMA_bInPotPosLeft); + writeb(bRight, dev->SMA + SMA_bInPotPosRight); + if (snd_msnd_send_word(dev, 0, 0, HDEXAR_IN_SET_POTS) == 0) + snd_msnd_send_dsp_cmd_chk(dev, HDEX_AUX_REQ); + break; +#ifndef MSND_CLASSIC + case MSND_MIXER_MIC: /* mic pot control */ + /* scaled by IMIX in digital mix */ + writeb(bLeft, dev->SMA + SMA_bMicPotPosLeft); + writeb(bRight, dev->SMA + SMA_bMicPotPosRight); + if (snd_msnd_send_word(dev, 0, 0, HDEXAR_MIC_SET_POTS) == 0) + snd_msnd_send_dsp_cmd_chk(dev, HDEX_AUX_REQ); + break; +#endif + case MSND_MIXER_VOLUME: /* master volume */ + writew(wLeft, dev->SMA + SMA_wCurrMastVolLeft); + writew(wRight, dev->SMA + SMA_wCurrMastVolRight); + /* fall through */ + + case MSND_MIXER_AUX: /* aux pot control */ + /* scaled by master volume */ + /* fall through */ + + /* digital controls */ + case MSND_MIXER_SYNTH: /* synth vol (dsp mix) */ + case MSND_MIXER_PCM: /* pcm vol (dsp mix) */ + case MSND_MIXER_IMIX: /* input monitor (dsp mix) */ + /* scaled by master volume */ + updatemaster = 1; + break; + + default: + return 0; + } + + if (updatemaster) { + /* update master volume scaled controls */ + update_volm(MSND_MIXER_PCM, wCurrPlayVol); + update_volm(MSND_MIXER_IMIX, wCurrInVol); +#ifndef MSND_CLASSIC + update_volm(MSND_MIXER_SYNTH, wCurrMHdrVol); +#endif + update_potm(MSND_MIXER_AUX, bAuxPotPos, HDEXAR_AUX_SET_POTS); + } + + return 0; +} + +static int snd_msndmix_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_msnd *msnd = snd_kcontrol_chip(kcontrol); + int change, addr = kcontrol->private_value; + int left, right; + unsigned long flags; + + left = ucontrol->value.integer.value[0] % 101; + right = ucontrol->value.integer.value[1] % 101; + spin_lock_irqsave(&msnd->mixer_lock, flags); + change = msnd->left_levels[addr] != left + || msnd->right_levels[addr] != right; + snd_msndmix_set(msnd, addr, left, right); + spin_unlock_irqrestore(&msnd->mixer_lock, flags); + return change; +} + + +#define DUMMY_VOLUME(xname, xindex, addr) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .info = snd_msndmix_volume_info, \ + .get = snd_msndmix_volume_get, .put = snd_msndmix_volume_put, \ + .private_value = addr } + + +static struct snd_kcontrol_new snd_msnd_controls[] = { +DUMMY_VOLUME("Master Volume", 0, MSND_MIXER_VOLUME), +DUMMY_VOLUME("PCM Volume", 0, MSND_MIXER_PCM), +DUMMY_VOLUME("Aux Volume", 0, MSND_MIXER_AUX), +DUMMY_VOLUME("Line Volume", 0, MSND_MIXER_LINE), +DUMMY_VOLUME("Mic Volume", 0, MSND_MIXER_MIC), +DUMMY_VOLUME("Monitor", 0, MSND_MIXER_IMIX), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = snd_msndmix_info_mux, + .get = snd_msndmix_get_mux, + .put = snd_msndmix_put_mux, +} +}; + + +int __devinit snd_msndmix_new(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + unsigned int idx; + int err; + + if (snd_BUG_ON(!chip)) + return -EINVAL; + spin_lock_init(&chip->mixer_lock); + strcpy(card->mixername, "MSND Pinnacle Mixer"); + + for (idx = 0; idx < ARRAY_SIZE(snd_msnd_controls); idx++) + err = snd_ctl_add(card, + snd_ctl_new1(snd_msnd_controls + idx, chip)); + if (err < 0) + return err; + + return 0; +} + +void snd_msndmix_setup(struct snd_msnd *dev) +{ + update_pot(MSND_MIXER_LINE, bInPotPos, HDEXAR_IN_SET_POTS); + update_potm(MSND_MIXER_AUX, bAuxPotPos, HDEXAR_AUX_SET_POTS); + update_volm(MSND_MIXER_PCM, wCurrPlayVol); + update_volm(MSND_MIXER_IMIX, wCurrInVol); +#ifndef MSND_CLASSIC + update_pot(MSND_MIXER_MIC, bMicPotPos, HDEXAR_MIC_SET_POTS); + update_volm(MSND_MIXER_SYNTH, wCurrMHdrVol); +#endif + +} + +unsigned long snd_msndmix_force_recsrc(struct snd_msnd *dev, int recsrc) +{ + dev->recsrc = -1; + return snd_msndmix_set_mux(dev, recsrc); +} + ---------------------------------------------------------------------- Sprawdz, co wyswietlaja kina w Twoim miescie! http://link.interia.pl/f2028

2 1

[alsa-devel] Turtle Beach Multisound Pinnacle alsa driver
by Krzysztof Helt 22 Jan '09

22 Jan '09

Hi, I got an access to old ISA card called Turtle Beach Multisound Pinnacle. I could not resist and update the Pinnacle driver from alsa-driver package to work with current kernel (2.6.28-rc6). The attached patch is current state of the driver. It still contains many formatting issues but it allows pcm and midi output (maybe input as well). This is for people who wants to try and report results. The driver requires firmware files put into firmware/turtlebeach directory. How to prepare these files is explained already in kernel: Documentation/sound/oss/MultiSound I made Multisound Classic driver compilable but I have no hardware to test. Main changes comparing to "old" driver from the alsa-driver package are: 1. Update probing framework to use ISA/PnP functions 2. Convert midi to use mpu401 driver (from rawmidi) 3. CS changes Regards, Krzysztof diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 660beb4..40f4a41 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -411,5 +411,36 @@ config SND_WAVEFRONT_FIRMWARE_IN_KERNEL you need to install the firmware files from the alsa-firmware package. +config SND_MSND_PINNACLE + tristate "Turtle Beach MultiSound Pinnacle/Fiji driver" + depends on X86 + select FW_LOADER + select SND_MPU401_UART + select SND_PCM + help + Say Y to include support for Turtle Beach MultiSound Pinnacle/ + Fiji soundcards. + + To compile this driver as a module, choose M here: the module + will be called snd-msnd-pinnacle. + +config SND_MSND_CLASSIC + tristate "Support for Turtle Beach MultiSound Classic, Tahiti, Monterey" + depends on X86 + select FW_LOADER + select SND_MPU401_UART + select SND_PCM + help + Say M here if you have a Turtle Beach MultiSound Classic, Tahiti or + Monterey (not for the Pinnacle or Fiji). + + See <file:Documentation/sound/oss/MultiSound> for important information + about this driver. Note that it has been discontinued, but the + Voyetra Turtle Beach knowledge base entry for it is still available + at <http://www.turtlebeach.com/site/kb_ftp/790.asp>. + + To compile this driver as a module, choose M here: the module + will be called snd-msnd-classic. + endif # SND_ISA diff --git a/sound/isa/Makefile b/sound/isa/Makefile index 63af13d..beb96ec 100644 --- a/sound/isa/Makefile +++ b/sound/isa/Makefile @@ -26,5 +26,5 @@ obj-$(CONFIG_SND_SC6000) += snd-sc6000.o obj-$(CONFIG_SND_SGALAXY) += snd-sgalaxy.o obj-$(CONFIG_SND_SSCAPE) += snd-sscape.o -obj-$(CONFIG_SND) += ad1816a/ ad1848/ cs423x/ es1688/ gus/ opti9xx/ \ - sb/ wavefront/ wss/ +obj-$(CONFIG_SND) += ad1816a/ ad1848/ cs423x/ es1688/ gus/ msnd/ \ + opti9xx/ sb/ wavefront/ wss/ diff --git a/sound/isa/msnd/Makefile b/sound/isa/msnd/Makefile new file mode 100644 index 0000000..5d3c814 --- /dev/null +++ b/sound/isa/msnd/Makefile @@ -0,0 +1,9 @@ + +snd-msnd-lib-objs := msnd.o msnd_midi.o +snd-msnd-pinnacle-objs := msnd_pinnacle.o msnd_pinnacle_mixer.o +snd-msnd-classic-objs := msnd_classic.o msnd_pinnacle_mixer.o + +# Toplevel Module Dependency +obj-$(CONFIG_SND_MSND_PINNACLE) += snd-msnd-pinnacle.o snd-msnd-lib.o +obj-$(CONFIG_SND_MSND_CLASSIC) += snd-msnd-classic.o snd-msnd-lib.o + diff --git a/sound/isa/msnd/msnd.c b/sound/isa/msnd/msnd.c new file mode 100644 index 0000000..cc17580 --- /dev/null +++ b/sound/isa/msnd/msnd.c @@ -0,0 +1,224 @@ +/********************************************************************* + * + * 2002/06/30 Karsten Wiese: + * removed kernel-version dependencies. + * ripped from linux kernel 2.4.18 (OSS Implementation) by me. + * In the OSS Version, this file is compiled to a separate MODULE, + * that is used by the pinnacle and the classic driver. + * since there is no classic driver for alsa yet (i dont have a classic + * & writing one blindfold is difficult) this file's object is statically + * linked into the pinnacle-driver-module for now. look for the string + * "uncomment this to make this a module again" + * to do guess what. + * + * the following is a copy of the 2.4.18 OSS FREE file-heading comment: + * + * msnd.c - Driver Base + * + * Turtle Beach MultiSound Sound Card Driver for Linux + * + * Copyright (C) 1998 Andrew Veliath + * + * 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., 675 Mass Ave, Cambridge, MA 02139, USA. + * + ********************************************************************/ + +#include <sound/core.h> +#include <sound/initval.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/slab.h> +#include <linux/types.h> +#include <linux/delay.h> + +#include <linux/interrupt.h> +#include <linux/io.h> +#include "msnd.h" + +#define LOGNAME "msnd" + + +void snd_msnd_init_queue(unsigned long base, int start, int size) +{ + isa_writew(PCTODSP_BASED(start), base + JQS_wStart); + isa_writew(PCTODSP_OFFSET(size) - 1, base + JQS_wSize); + isa_writew(0, base + JQS_wHead); + isa_writew(0, base + JQS_wTail); +} +EXPORT_SYMBOL(snd_msnd_init_queue); + + +int snd_msnd_wait_TXDE(struct snd_msnd *dev) +{ + unsigned int io = dev->io; + int timeout = 1000; + + while (timeout-- > 0) + if (inb(io + HP_ISR) & HPISR_TXDE) + return 0; + + return -EIO; +} +EXPORT_SYMBOL(snd_msnd_wait_TXDE); + +int snd_msnd_wait_HC0(struct snd_msnd *dev) +{ + unsigned int io = dev->io; + int timeout = 1000; + + while (timeout-- > 0) + if (!(inb(io + HP_CVR) & HPCVR_HC)) + return 0; + + return -EIO; +} +EXPORT_SYMBOL(snd_msnd_wait_HC0); + +int snd_msnd_send_dsp_cmd(struct snd_msnd *dev, u8 cmd) +{ + unsigned long flags; + + spin_lock_irqsave(&dev->lock, flags); + if (snd_msnd_wait_HC0(dev) == 0) { + outb(cmd, dev->io + HP_CVR); + spin_unlock_irqrestore(&dev->lock, flags); + return 0; + } + spin_unlock_irqrestore(&dev->lock, flags); + + printk(KERN_DEBUG LOGNAME ": Send DSP command timeout\n"); + + return -EIO; +} +EXPORT_SYMBOL(snd_msnd_send_dsp_cmd); + + +int snd_msnd_send_word(struct snd_msnd *dev, unsigned char high, + unsigned char mid, unsigned char low) +{ + unsigned int io = dev->io; + + if (snd_msnd_wait_TXDE(dev) == 0) { + outb(high, io + HP_TXH); + outb(mid, io + HP_TXM); + outb(low, io + HP_TXL); + return 0; + } + + printk(KERN_DEBUG LOGNAME ": Send host word timeout\n"); + + return -EIO; +} +EXPORT_SYMBOL(snd_msnd_send_word); + +int snd_msnd_upload_host(struct snd_msnd *dev, const u8 *bin, int len) +{ + int i; + + if (len % 3 != 0) { + printk(KERN_WARNING + LOGNAME ": Upload host data not multiple of 3!\n"); + return -EINVAL; + } + + for (i = 0; i < len; i += 3) + if (snd_msnd_send_word(dev, bin[i], bin[i + 1], bin[i + 2]) != 0) + return -EIO; + + inb(dev->io + HP_RXL); + inb(dev->io + HP_CVR); + + return 0; +} +EXPORT_SYMBOL(snd_msnd_upload_host); + +int snd_msnd_enable_irq(struct snd_msnd *dev) +{ + unsigned long flags; + + if (dev->irq_ref++) + return 0; + + printk(KERN_DEBUG LOGNAME ": Enabling IRQ\n"); + + spin_lock_irqsave(&dev->lock, flags); + if (snd_msnd_wait_TXDE(dev) == 0) { + outb(inb(dev->io + HP_ICR) | HPICR_TREQ, dev->io + HP_ICR); + if (dev->type == msndClassic) + outb(dev->irqid, dev->io + HP_IRQM); + + outb(inb(dev->io + HP_ICR) & ~HPICR_TREQ, dev->io + HP_ICR); + outb(inb(dev->io + HP_ICR) | HPICR_RREQ, dev->io + HP_ICR); + enable_irq(dev->irq); + snd_msnd_init_queue(dev->DSPQ, dev->dspq_data_buff, + dev->dspq_buff_size); + spin_unlock_irqrestore(&dev->lock, flags); + return 0; + } + spin_unlock_irqrestore(&dev->lock, flags); + + printk(KERN_DEBUG LOGNAME ": Enable IRQ failed\n"); + + return -EIO; +} +EXPORT_SYMBOL(snd_msnd_enable_irq); + + +int snd_msnd_disable_irq(struct snd_msnd *dev) +{ + unsigned long flags; + + if (--dev->irq_ref > 0) + return 0; + + if (dev->irq_ref < 0) + printk(KERN_DEBUG LOGNAME ": IRQ ref count is %d\n", + dev->irq_ref); + + printk(KERN_DEBUG LOGNAME ": Disabling IRQ\n"); + + spin_lock_irqsave(&dev->lock, flags); + if (snd_msnd_wait_TXDE(dev) == 0) { + outb(inb(dev->io + HP_ICR) & ~HPICR_RREQ, dev->io + HP_ICR); + if (dev->type == msndClassic) + outb(HPIRQ_NONE, dev->io + HP_IRQM); + disable_irq(dev->irq); + spin_unlock_irqrestore(&dev->lock, flags); + return 0; + } + spin_unlock_irqrestore(&dev->lock, flags); + + printk(KERN_DEBUG LOGNAME ": Disable IRQ failed\n"); + + return -EIO; +} +EXPORT_SYMBOL(snd_msnd_disable_irq); + + +MODULE_AUTHOR("Andrew Veliath <andrewtv(a)usa.net>"); +MODULE_DESCRIPTION("Turtle Beach MultiSound Driver Base"); +MODULE_LICENSE("GPL"); + + +int init_module(void) +{ + return 0; +} + +void cleanup_module(void) +{ +} + diff --git a/sound/isa/msnd/msnd.h b/sound/isa/msnd/msnd.h new file mode 100644 index 0000000..7d07ce6 --- /dev/null +++ b/sound/isa/msnd/msnd.h @@ -0,0 +1,341 @@ +/********************************************************************* + * + * msnd.h + * + * Turtle Beach MultiSound Sound Card Driver for Linux + * + * Some parts of this header file were derived from the Turtle Beach + * MultiSound Driver Development Kit. + * + * Copyright (C) 1998 Andrew Veliath + * Copyright (C) 1993 Turtle Beach Systems, Inc. + * + * 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., 675 Mass Ave, Cambridge, MA 02139, USA. + * + ********************************************************************/ +#ifndef __MSND_H +#define __MSND_H + +#define VERSION "0.1.3.0" + +#define DEFSAMPLERATE 44100 +#define DEFSAMPLESIZE SNDRV_PCM_FORMAT_S16 +#define DEFCHANNELS 1 + +#define SNDCARD_MSND 38 + +#define SRAM_BANK_SIZE 0x8000 +#define SRAM_CNTL_START 0x7F00 + +#define DSP_BASE_ADDR 0x4000 +#define DSP_BANK_BASE 0x4000 + +#define HP_ICR 0x00 +#define HP_CVR 0x01 +#define HP_ISR 0x02 +#define HP_IVR 0x03 +#define HP_NU 0x04 +#define HP_INFO 0x04 +#define HP_TXH 0x05 +#define HP_RXH 0x05 +#define HP_TXM 0x06 +#define HP_RXM 0x06 +#define HP_TXL 0x07 +#define HP_RXL 0x07 + +#define HP_ICR_DEF 0x00 +#define HP_CVR_DEF 0x12 +#define HP_ISR_DEF 0x06 +#define HP_IVR_DEF 0x0f +#define HP_NU_DEF 0x00 + +#define HP_IRQM 0x09 + +#define HPR_BLRC 0x08 +#define HPR_SPR1 0x09 +#define HPR_SPR2 0x0A +#define HPR_TCL0 0x0B +#define HPR_TCL1 0x0C +#define HPR_TCL2 0x0D +#define HPR_TCL3 0x0E +#define HPR_TCL4 0x0F + +#define HPICR_INIT 0x80 +#define HPICR_HM1 0x40 +#define HPICR_HM0 0x20 +#define HPICR_HF1 0x10 +#define HPICR_HF0 0x08 +#define HPICR_TREQ 0x02 +#define HPICR_RREQ 0x01 + +#define HPCVR_HC 0x80 + +#define HPISR_HREQ 0x80 +#define HPISR_DMA 0x40 +#define HPISR_HF3 0x10 +#define HPISR_HF2 0x08 +#define HPISR_TRDY 0x04 +#define HPISR_TXDE 0x02 +#define HPISR_RXDF 0x01 + +#define HPIO_290 0 +#define HPIO_260 1 +#define HPIO_250 2 +#define HPIO_240 3 +#define HPIO_230 4 +#define HPIO_220 5 +#define HPIO_210 6 +#define HPIO_3E0 7 + +#define HPMEM_NONE 0 +#define HPMEM_B000 1 +#define HPMEM_C800 2 +#define HPMEM_D000 3 +#define HPMEM_D400 4 +#define HPMEM_D800 5 +#define HPMEM_E000 6 +#define HPMEM_E800 7 + +#define HPIRQ_NONE 0 +#define HPIRQ_5 1 +#define HPIRQ_7 2 +#define HPIRQ_9 3 +#define HPIRQ_10 4 +#define HPIRQ_11 5 +#define HPIRQ_12 6 +#define HPIRQ_15 7 + +#define HIMT_PLAY_DONE 0x00 +#define HIMT_RECORD_DONE 0x01 +#define HIMT_MIDI_EOS 0x02 +#define HIMT_MIDI_OUT 0x03 + +#define HIMT_MIDI_IN_UCHAR 0x0E +#define HIMT_DSP 0x0F + +#define HDEX_BASE 0x92 +#define HDEX_PLAY_START (0 + HDEX_BASE) +#define HDEX_PLAY_STOP (1 + HDEX_BASE) +#define HDEX_PLAY_PAUSE (2 + HDEX_BASE) +#define HDEX_PLAY_RESUME (3 + HDEX_BASE) +#define HDEX_RECORD_START (4 + HDEX_BASE) +#define HDEX_RECORD_STOP (5 + HDEX_BASE) +#define HDEX_MIDI_IN_START (6 + HDEX_BASE) +#define HDEX_MIDI_IN_STOP (7 + HDEX_BASE) +#define HDEX_MIDI_OUT_START (8 + HDEX_BASE) +#define HDEX_MIDI_OUT_STOP (9 + HDEX_BASE) +#define HDEX_AUX_REQ (10 + HDEX_BASE) + +#define HIWORD(l) ((u16)((((u32)(l)) >> 16) & 0xFFFF)) +#define LOWORD(l) ((u16)(u32)(l)) +#define HIBYTE(w) ((u8)(((u16)(w) >> 8) & 0xFF)) +#define LOBYTE(w) ((u8)(w)) +#define MAKELONG(low, hi) ((long)(((u16)(low))|(((u32)((u16)(hi)))<<16))) +#define MAKEWORD(low, hi) ((u16)(((u8)(low))|(((u16)((u8)(hi)))<<8))) + +#define PCTODSP_OFFSET(w) (u16)((w)/2) +#define PCTODSP_BASED(w) (u16)(((w)/2) + DSP_BASE_ADDR) +#define DSPTOPC_BASED(w) (((w) - DSP_BASE_ADDR) * 2) + +#ifdef SLOWIO +# undef outb +# undef inb +# define outb outb_p +# define inb inb_p +#endif + +/* JobQueueStruct */ +#define JQS_wStart 0x00 +#define JQS_wSize 0x02 +#define JQS_wHead 0x04 +#define JQS_wTail 0x06 +#define JQS__size 0x08 + +/* DAQueueDataStruct */ +#define DAQDS_wStart 0x00 +#define DAQDS_wSize 0x02 +#define DAQDS_wFormat 0x04 +#define DAQDS_wSampleSize 0x06 +#define DAQDS_wChannels 0x08 +#define DAQDS_wSampleRate 0x0A +#define DAQDS_wIntMsg 0x0C +#define DAQDS_wFlags 0x0E +#define DAQDS__size 0x10 + +/* Generic FIFO * / +typedef struct { + size_t n, len; + char *data; + int head, tail; +} msnd_fifo; */ + + + +struct snd_msnd { + void *mappedbase; + int play_period_bytes; + int playLimit; + int playPeriods; + int playDMAPos; + int captureDMAPos; + int capturePeriodBytes; + int captureLimit; + int capturePeriods; + struct snd_card *card; + void *msndmidi_mpu; + struct snd_rawmidi *rmidi; + + /* Hardware resources */ + long io; + int memid, irqid; + int irq, irq_ref; + unsigned long base; + + /* Motorola 56k DSP SMA */ + unsigned long SMA; + unsigned long DAPQ, DARQ, MODQ, MIDQ, DSPQ; + unsigned long pwDSPQData, pwMIDQData, pwMODQData; + int dspq_data_buff, dspq_buff_size; + + /* State variables */ + enum { msndClassic, msndPinnacle } type; + mode_t mode; + unsigned long flags; +#define F_RESETTING 0 +#define F_HAVEDIGITAL 1 +#define F_AUDIO_WRITE_INUSE 2 +#define F_WRITING 3 +#define F_WRITEBLOCK 4 +#define F_WRITEFLUSH 5 +#define F_AUDIO_READ_INUSE 6 +#define F_READING 7 +#define F_READBLOCK 8 +#define F_EXT_MIDI_INUSE 9 +#define F_HDR_MIDI_INUSE 10 +#define F_DISABLE_WRITE_NDELAY 11 + spinlock_t lock; + int nresets; + unsigned recsrc; +#define LEVEL_ENTRIES 32 + int left_levels[LEVEL_ENTRIES]; + int right_levels[LEVEL_ENTRIES]; + int mixer_mod_count; + int calibrate_signal; + int play_sample_size, play_sample_rate, play_channels; + int play_ndelay; + int capture_sample_size, capture_sample_rate, capture_channels; + int capture_ndelay; + u8 bCurrentMidiPatch; + + int last_playbank, last_recbank; + struct snd_pcm_substream *playback_substream; + struct snd_pcm_substream *capture_substream; + +}; + +void snd_msnd_init_queue(unsigned long, int start, int size); + +int snd_msnd_wait_TXDE(struct snd_msnd *chip); +int snd_msnd_wait_HC0(struct snd_msnd *chip); +int snd_msnd_send_dsp_cmd(struct snd_msnd *chip, u8 cmd); +int snd_msnd_send_word(struct snd_msnd *chip, + unsigned char high, + unsigned char mid, + unsigned char low); +int snd_msnd_upload_host(struct snd_msnd *chip, + const u8 *bin, int len); +int snd_msnd_enable_irq(struct snd_msnd *chip); +int snd_msnd_disable_irq(struct snd_msnd *chip); + + +int snd_msndmidi_new(struct snd_card *card, int device); +void snd_msndmidi_input_read(void *mpu); + +static inline u8 +isa_readb(unsigned long offset) +{ + void __iomem *addr = ioremap(offset, 1); + u8 ret = readb(addr); + iounmap(addr); + return ret; +} + +static inline u16 +isa_readw(unsigned long offset) +{ + void __iomem *addr = ioremap(offset, 2); + u16 ret = readw(addr); + iounmap(addr); + return ret; +} + +static inline u32 +isa_readl(unsigned long offset) +{ + void __iomem *addr = ioremap(offset, 4); + u32 ret = readl(addr); + iounmap(addr); + return ret; +} + +static inline void +isa_writeb(u8 b, unsigned long offset) +{ + void __iomem *addr = ioremap(offset, 2); + writeb(b, addr); + iounmap(addr); +} + +static inline void +isa_writew(u16 w, unsigned long offset) +{ + void __iomem *addr = ioremap(offset, 2); + writew(w, addr); + iounmap(addr); +} + +static inline void +isa_writel(u32 l, unsigned long offset) +{ + void __iomem *addr = ioremap(offset, 4); + writel(l, addr); + iounmap(addr); +} + +static inline void +isa_memset_io(unsigned long offset, u8 val, long n) +{ + void __iomem *addr = ioremap(offset, n); + memset_io(addr, val, n); + iounmap(addr); +} + +static inline void +isa_memcpy_fromio(void *dest, unsigned long offset, long n) +{ + void __iomem *addr = ioremap(offset, n); + memcpy_fromio(dest, addr, n); + iounmap(addr); +} + +static inline void +isa_memcpy_toio(unsigned long offset, const void *src, long n) +{ + void __iomem *addr = ioremap(offset, n); + memcpy_toio(addr, src, n); + iounmap(addr); +} + +#endif /* __MSND_H */ diff --git a/sound/isa/msnd/msnd_classic.c b/sound/isa/msnd/msnd_classic.c new file mode 100644 index 0000000..3b23a09 --- /dev/null +++ b/sound/isa/msnd/msnd_classic.c @@ -0,0 +1,3 @@ +/* The work is in msnd_pinnacle.c, just define MSND_CLASSIC before it. */ +#define MSND_CLASSIC +#include "msnd_pinnacle.c" diff --git a/sound/isa/msnd/msnd_classic.h b/sound/isa/msnd/msnd_classic.h new file mode 100644 index 0000000..f4f2cc9 --- /dev/null +++ b/sound/isa/msnd/msnd_classic.h @@ -0,0 +1,182 @@ +/********************************************************************* + * + * msnd_classic.h + * + * Turtle Beach MultiSound Sound Card Driver for Linux + * + * Some parts of this header file were derived from the Turtle Beach + * MultiSound Driver Development Kit. + * + * Copyright (C) 1998 Andrew Veliath + * Copyright (C) 1993 Turtle Beach Systems, Inc. + * + * 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., 675 Mass Ave, Cambridge, MA 02139, USA. + * + ********************************************************************/ +#ifndef __MSND_CLASSIC_H +#define __MSND_CLASSIC_H + +#define DSP_NUMIO 0x10 + +#define HP_MEMM 0x08 + +#define HP_BITM 0x0E +#define HP_WAIT 0x0D +#define HP_DSPR 0x0A +#define HP_PROR 0x0B +#define HP_BLKS 0x0C + +#define HPPRORESET_OFF 0 +#define HPPRORESET_ON 1 + +#define HPDSPRESET_OFF 0 +#define HPDSPRESET_ON 1 + +#define HPBLKSEL_0 0 +#define HPBLKSEL_1 1 + +#define HPWAITSTATE_0 0 +#define HPWAITSTATE_1 1 + +#define HPBITMODE_16 0 +#define HPBITMODE_8 1 + +#define HIDSP_INT_PLAY_UNDER 0x00 +#define HIDSP_INT_RECORD_OVER 0x01 +#define HIDSP_INPUT_CLIPPING 0x02 +#define HIDSP_MIDI_IN_OVER 0x10 +#define HIDSP_MIDI_OVERRUN_ERR 0x13 + +#define HDEXAR_CLEAR_PEAKS 1 +#define HDEXAR_IN_SET_POTS 2 +#define HDEXAR_AUX_SET_POTS 3 +#define HDEXAR_CAL_A_TO_D 4 +#define HDEXAR_RD_EXT_DSP_BITS 5 + +#define TIME_PRO_RESET_DONE 0x028A +#define TIME_PRO_SYSEX 0x0040 +#define TIME_PRO_RESET 0x0032 + +#define AGND 0x01 +#define SIGNAL 0x02 + +#define EXT_DSP_BIT_DCAL 0x0001 +#define EXT_DSP_BIT_MIDI_CON 0x0002 + +#define BUFFSIZE 0x8000 +#define HOSTQ_SIZE 0x40 + +#define SRAM_CNTL_START 0x7F00 +#define SMA_STRUCT_START 0x7F40 + +#define DAP_BUFF_SIZE 0x2400 +#define DAR_BUFF_SIZE 0x2000 + +#define DAPQ_STRUCT_SIZE 0x10 +#define DARQ_STRUCT_SIZE 0x10 +#define DAPQ_BUFF_SIZE (3 * 0x10) +#define DARQ_BUFF_SIZE (3 * 0x10) +#define MODQ_BUFF_SIZE 0x400 +#define MIDQ_BUFF_SIZE 0x200 +#define DSPQ_BUFF_SIZE 0x40 + +#define DAPQ_DATA_BUFF 0x6C00 +#define DARQ_DATA_BUFF 0x6C30 +#define MODQ_DATA_BUFF 0x6C60 +#define MIDQ_DATA_BUFF 0x7060 +#define DSPQ_DATA_BUFF 0x7260 + +#define DAPQ_OFFSET SRAM_CNTL_START +#define DARQ_OFFSET (SRAM_CNTL_START + 0x08) +#define MODQ_OFFSET (SRAM_CNTL_START + 0x10) +#define MIDQ_OFFSET (SRAM_CNTL_START + 0x18) +#define DSPQ_OFFSET (SRAM_CNTL_START + 0x20) + +#define MOP_SYNTH 0x10 +#define MOP_EXTOUT 0x32 +#define MOP_EXTTHRU 0x02 +#define MOP_OUTMASK 0x01 + +#define MIP_EXTIN 0x01 +#define MIP_SYNTH 0x00 +#define MIP_INMASK 0x32 + +/* Classic SMA Common Data */ +#define SMA_wCurrPlayBytes 0x0000 +#define SMA_wCurrRecordBytes 0x0002 +#define SMA_wCurrPlayVolLeft 0x0004 +#define SMA_wCurrPlayVolRight 0x0006 +#define SMA_wCurrInVolLeft 0x0008 +#define SMA_wCurrInVolRight 0x000a +#define SMA_wUser_3 0x000c +#define SMA_wUser_4 0x000e +#define SMA_dwUser_5 0x0010 +#define SMA_dwUser_6 0x0014 +#define SMA_wUser_7 0x0018 +#define SMA_wReserved_A 0x001a +#define SMA_wReserved_B 0x001c +#define SMA_wReserved_C 0x001e +#define SMA_wReserved_D 0x0020 +#define SMA_wReserved_E 0x0022 +#define SMA_wReserved_F 0x0024 +#define SMA_wReserved_G 0x0026 +#define SMA_wReserved_H 0x0028 +#define SMA_wCurrDSPStatusFlags 0x002a +#define SMA_wCurrHostStatusFlags 0x002c +#define SMA_wCurrInputTagBits 0x002e +#define SMA_wCurrLeftPeak 0x0030 +#define SMA_wCurrRightPeak 0x0032 +#define SMA_wExtDSPbits 0x0034 +#define SMA_bExtHostbits 0x0036 +#define SMA_bBoardLevel 0x0037 +#define SMA_bInPotPosRight 0x0038 +#define SMA_bInPotPosLeft 0x0039 +#define SMA_bAuxPotPosRight 0x003a +#define SMA_bAuxPotPosLeft 0x003b +#define SMA_wCurrMastVolLeft 0x003c +#define SMA_wCurrMastVolRight 0x003e +#define SMA_bUser_12 0x0040 +#define SMA_bUser_13 0x0041 +#define SMA_wUser_14 0x0042 +#define SMA_wUser_15 0x0044 +#define SMA_wCalFreqAtoD 0x0046 +#define SMA_wUser_16 0x0048 +#define SMA_wUser_17 0x004a +#define SMA__size 0x004c + +#ifdef HAVE_DSPCODEH +# include "msndperm.c" +# include "msndinit.c" +# define PERMCODE msndperm +# define INITCODE msndinit +# define PERMCODESIZE sizeof(msndperm) +# define INITCODESIZE sizeof(msndinit) +#else +# ifndef CONFIG_MSNDCLAS_INIT_FILE +# define CONFIG_MSNDCLAS_INIT_FILE "turtlebeach/msndinit.bin" +# endif +# ifndef CONFIG_MSNDCLAS_PERM_FILE +# define CONFIG_MSNDCLAS_PERM_FILE "turtlebeach/msndperm.bin" +# endif +# define PERMCODEFILE CONFIG_MSNDCLAS_PERM_FILE +# define INITCODEFILE CONFIG_MSNDCLAS_INIT_FILE +# define PERMCODE dspini +# define INITCODE permini +# define PERMCODESIZE sizeof_dspini +# define INITCODESIZE sizeof_permini +#endif +#define LONGNAME "MultiSound (Classic/Monterey/Tahiti)" + +#endif /* __MSND_CLASSIC_H */ diff --git a/sound/isa/msnd/msnd_midi.c b/sound/isa/msnd/msnd_midi.c new file mode 100644 index 0000000..99feb40 --- /dev/null +++ b/sound/isa/msnd/msnd_midi.c @@ -0,0 +1,385 @@ +/* + * Copyright (c) by Jaroslav Kysela <perex(a)perex.cz> + * Routines for control of MPU-401 in UART mode + * + * MPU-401 supports UART mode which is not capable generate transmit + * interrupts thus output is done via polling. Also, if irq < 0, then + * input is done also via polling. Do not expect good performance. + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <linux/io.h> +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/ioport.h> +#include <linux/sched.h> +#include <linux/errno.h> +#include <sound/core.h> +#include <sound/rawmidi.h> + +#include "msnd.h" +#ifdef MSND_CLASSIC +# ifdef CONFIG_MSNDCLAS_HAVE_BOOT +# define HAVE_DSPCODEH +# endif +# include "msnd_classic.h" +# define LOGNAME "msnd_classic" +#else +# ifdef CONFIG_MSNDPIN_HAVE_BOOT +# define HAVE_DSPCODEH +# endif +# include "msnd_pinnacle.h" +# define LOGNAME "snd_msnd_pinnacle" +#endif + + +#define MSNDMIDI_MODE_BIT_INPUT 0 +#define MSNDMIDI_MODE_BIT_OUTPUT 1 +#define MSNDMIDI_MODE_BIT_INPUT_TRIGGER 2 +#define MSNDMIDI_MODE_BIT_OUTPUT_TRIGGER 3 + +#define MSNDMIDI_MODE_INPUT (1<<MSNDMIDI_MODE_BIT_INPUT) +#define MSNDMIDI_MODE_OUTPUT (1<<MSNDMIDI_MODE_BIT_OUTPUT) +#define MSNDMIDI_MODE_INPUT_TRIGGER (1<<MSNDMIDI_MODE_BIT_INPUT_TRIGGER) +#define MSNDMIDI_MODE_OUTPUT_TRIGGER (1<<MSNDMIDI_MODE_BIT_OUTPUT_TRIGGER) + +#define MSNDMIDI_MODE_INPUT_TIMER (1<<0) +#define MSNDMIDI_MODE_OUTPUT_TIMER (1<<1) + + +/* + + */ + +#define snd_msndmidi_input_avail(mpu) (!(inb(MPU401C(mpu)) & 0x80)) +#define snd_msndmidi_output_ready(mpu) (!(inb(MPU401C(mpu)) & 0x40)) + +#define MPU401_RESET 0xff +#define MPU401_ENTER_UART 0x3f +#define MPU401_ACK 0xfe + + +struct snd_msndmidi { + struct snd_rawmidi *rmidi; + struct snd_msnd *dev; + + unsigned long mode; /* MSNDMIDI_MODE_XXXX */ + int timer_invoked; + + void *private_data; + + struct snd_rawmidi_substream *substream_input; + struct snd_rawmidi_substream *substream_output; + + spinlock_t input_lock; + spinlock_t output_lock; + spinlock_t timer_lock; + + struct timer_list timer; +}; + + +/* not used static void snd_msnd_midi_clear_rx(struct msndmidi *mpu) +{/ * int timeout = 100000; + for (; timeout > 0 && snd_msndmidi_input_avail(mpu); timeout--) + inb(MPU401D(mpu)); +#ifdef CONFIG_SND_DEBUG + if (timeout <= 0) + snd_printk("cmd: clear rx timeout (status = 0x%x)\n", inb(MPU401C(mpu))); +#endif +* / +} */ + +/* +static void _snd_msndmidi_interrupt(struct msndmidi *mpu) +{ + if (test_bit(MPU401_MODE_BIT_INPUT, &mpu->mode)) + snd_msndmidi_input_read(mpu); + else + snd_msndmidi_clear_rx(mpu); + / * ok. for better Tx performance try do some output when input is done * / + if (test_bit(MPU401_MODE_BIT_OUTPUT, &mpu->mode)) + snd_msndmidi_output_write(mpu); +} + +void snd_msndmidi_interrupt(int irq, void *dev_id, struct pt_regs *regs) +{ + struct msndmidi *mpu = dev_id; + + if (mpu == NULL) + return; + _snd_msndmidi_interrupt(mpu); +}*/ +/* +static void snd_msndmidi_timer(unsigned long data) +{ + unsigned long flags; + struct msndmidi *mpu = (void *)data; + + spin_lock_irqsave(&mpu->timer_lock, flags); + / * mpu->mode |= MPU401_MODE_TIMER;* / + mpu->timer.expires = 1 + jiffies; + add_timer(&mpu->timer); + spin_unlock_irqrestore(&mpu->timer_lock, flags); + if (mpu->rmidi) + _snd_msndmidi_interrupt(mpu); +} + +static void snd_msndmidi_add_timer (struct msndmidi *mpu, int input) +{ + unsigned long flags; + + spin_lock_irqsave (&mpu->timer_lock, flags); + if (mpu->timer_invoked == 0) { + mpu->timer.data = (unsigned long)mpu; + mpu->timer.function = snd_msndmidi_timer; + mpu->timer.expires = 1 + jiffies; + add_timer(&mpu->timer); + } + mpu->timer_invoked |= input ? MPU401_MODE_INPUT_TIMER : MPU401_MODE_OUTPUT_TIMER; + spin_unlock_irqrestore (&mpu->timer_lock, flags); +} + +static void snd_msndmidi_remove_timer (struct snd_msndmidi *mpu, int input) +{ + unsigned long flags; + + spin_lock_irqsave (&mpu->timer_lock, flags); + if (mpu->timer_invoked) { + mpu->timer_invoked &= input ? ~MPU401_MODE_INPUT_TIMER : ~MPU401_MODE_OUTPUT_TIMER; + if (!mpu->timer_invoked) + del_timer(&mpu->timer); + } + spin_unlock_irqrestore (&mpu->timer_lock, flags); +} */ + +/* + + */ + +/* not used so far .... ? +static void snd_msndmidi_cmd(struct snd_msndmidi *mpu, unsigned char cmd, int ack) +{ + unsigned long flags; + int timeout, ok; + + spin_lock_irqsave(&mpu->input_lock, flags); +/ * if (mpu->hardware != MPU401_HW_TRID4DWAVE) { + outb(0x00, MPU401D(mpu)); + / * snd_msndmidi_clear_rx(mpu);* / + } + / * ok. standard MPU-401 initialization * / + if (mpu->hardware != MPU401_HW_SB) { + for (timeout = 1000; timeout > 0 && !snd_msndmidi_output_ready(mpu); timeout--) + udelay(10); +#ifdef CONFIG_SND_DEBUG + if (!timeout) + snd_printk("cmd: tx timeout (status = 0x%x)\n", inb(MPU401C(mpu))); +#endif + } + outb(cmd, MPU401C(mpu)); + if (ack) { + ok = 0; + timeout = 10000; + while (!ok && timeout-- > 0) { + if (snd_msndmidi_input_avail(mpu)) { + if (inb(MPU401D(mpu)) == MPU401_ACK) + ok = 1; + } + } + if (!ok && inb(MPU401D(mpu)) == MPU401_ACK) + ok = 1; + } else { + ok = 1; + }* / + spin_unlock_irqrestore(&mpu->input_lock, flags); +// if (!ok) +// snd_printk("cmd: 0x%x failed at 0x%lx (status = 0x%x, data = 0x%x)\n", cmd, mpu->port, inb(MPU401C(mpu)), inb(MPU401D(mpu))); + // snd_printk("cmd: 0x%x at 0x%lx (status = 0x%x, data = 0x%x)\n", cmd, mpu->port, inb(MPU401C(mpu)), inb(MPU401D(mpu))); +} */ + +/* + * input/output open/close - protected by open_mutex in rawmidi.c + */ +static int snd_msndmidi_input_open(struct snd_rawmidi_substream *substream) +{ + struct snd_msndmidi *mpu; +// int err; +#ifdef CONFIG_SND_DEBUG0 + printk(KERN_DEBUG "snd_msndmidi_input_open(struct snd_rawmidi_substream *substream)\n"); +#endif + + mpu = substream->rmidi->private_data; +/* if (mpu->open_input && (err = mpu->open_input(mpu)) < 0) + return err; + if (!test_bit(MPU401_MODE_BIT_OUTPUT, &mpu->mode)) { + snd_msndmidi_cmd(mpu, MPU401_RESET, 1); + snd_msndmidi_cmd(mpu, MPU401_ENTER_UART, 1); + }*/ + + mpu->substream_input = substream; + + // SetMidiInPort(EXTIN_MIP); + snd_msnd_enable_irq(mpu->dev); + + snd_msnd_send_dsp_cmd(mpu->dev, HDEX_MIDI_IN_START); + set_bit(MSNDMIDI_MODE_BIT_INPUT, &mpu->mode); + return 0; +} + +static int snd_msndmidi_input_close(struct snd_rawmidi_substream *substream) +{ + struct snd_msndmidi *mpu; + +#ifdef CONFIG_SND_DEBUG0 + printk(KERN_DEBUG "snd_msndmidi_input_close(struct snd_rawmidi_substream *substream)\n"); +#endif + + mpu = substream->rmidi->private_data; + snd_msnd_send_dsp_cmd(mpu->dev, HDEX_MIDI_IN_STOP); + clear_bit(MSNDMIDI_MODE_BIT_INPUT, &mpu->mode); + mpu->substream_input = NULL; + snd_msnd_disable_irq(mpu->dev); +/* if (!test_bit(MPU401_MODE_BIT_OUTPUT, &mpu->mode)) + snd_msndmidi_cmd(mpu, MPU401_RESET, 0); + if (mpu->close_input) + mpu->close_input(mpu);*/ + return 0; +} + +static void snd_msndmidi_input_drop(struct snd_msndmidi *mpu) +{ + u16 tail; + + tail = isa_readw(mpu->dev->MIDQ + JQS_wTail); + isa_writew(tail, mpu->dev->MIDQ + JQS_wHead); +} + +/* + * trigger input + */ +static void snd_msndmidi_input_trigger(struct snd_rawmidi_substream *substream, int up) +{ + unsigned long flags; + struct snd_msndmidi *mpu; +// int max = 64; +#ifdef CONFIG_SND_DEBUG0 + printk(KERN_DEBUG "snd_msndmidi_input_trigger(, %i)\n", up); +#endif + + mpu = substream->rmidi->private_data; + spin_lock_irqsave(&mpu->input_lock, flags); + if (up) { + if (!test_and_set_bit(MSNDMIDI_MODE_BIT_INPUT_TRIGGER, &mpu->mode)) + snd_msndmidi_input_drop(mpu); + } else { + clear_bit(MSNDMIDI_MODE_BIT_INPUT_TRIGGER, &mpu->mode); + } + spin_unlock_irqrestore(&mpu->input_lock, flags); + if (up) + snd_msndmidi_input_read(mpu); +} + +void snd_msndmidi_input_read(void *mpuv) +{ + unsigned long flags; + struct snd_msndmidi *mpu = mpuv; + + spin_lock_irqsave(&mpu->input_lock, flags); + while (isa_readw(mpu->dev->MIDQ + JQS_wTail) != isa_readw(mpu->dev->MIDQ + JQS_wHead)) { + u16 wTmp, val; + val = isa_readw(mpu->dev->pwMIDQData + 2*isa_readw(mpu->dev->MIDQ + JQS_wHead)); + + if (test_bit(MSNDMIDI_MODE_BIT_INPUT_TRIGGER, &mpu->mode)) { + // printk("MID: 0x%04X\n", (unsigned)val); + snd_rawmidi_receive(mpu->substream_input, (unsigned char *)&val, 1); + } + + wTmp = isa_readw(mpu->dev->MIDQ + JQS_wHead) + 1; + if (wTmp > isa_readw(mpu->dev->MIDQ + JQS_wSize)) + isa_writew(0, mpu->dev->MIDQ + JQS_wHead); + else + isa_writew(wTmp, mpu->dev->MIDQ + JQS_wHead); + } + spin_unlock_irqrestore(&mpu->input_lock, flags); +} +EXPORT_SYMBOL(snd_msndmidi_input_read); + +static struct snd_rawmidi_ops snd_msndmidi_input = { + .open = snd_msndmidi_input_open, + .close = snd_msndmidi_input_close, + .trigger = snd_msndmidi_input_trigger, +}; + +static void snd_msndmidi_free(struct snd_rawmidi *rmidi) +{ + struct snd_msndmidi *mpu = rmidi->private_data; +/* if (mpu->irq_flags && mpu->irq >= 0) + free_irq(mpu->irq, (void *) mpu); + if (mpu->res) { + release_resource(mpu->res); + kfree_nocheck(mpu->res); + }*/ + kfree(mpu); +} + +int snd_msndmidi_new(struct snd_card *card, int device) +{ + struct snd_msnd *chip = card->private_data; + struct snd_msndmidi *mpu; + struct snd_rawmidi *rmidi; + int err; + + err = snd_rawmidi_new(card, "MSND-MIDI", device, 1, 1, &rmidi); + if (err < 0) + return err; + mpu = kcalloc(1, sizeof(*mpu), GFP_KERNEL); + if (mpu == NULL) { + snd_device_free(card, rmidi); + return -ENOMEM; + } + mpu->dev = chip; + chip->msndmidi_mpu = mpu; + rmidi->private_data = mpu; + rmidi->private_free = snd_msndmidi_free; + spin_lock_init(&mpu->input_lock); + spin_lock_init(&mpu->output_lock); + spin_lock_init(&mpu->timer_lock); +/* if (!integrated) { + if ((mpu->res = request_region(port, 2, "MPU401 UART")) == NULL) { + snd_device_free(card, rmidi); + return -EBUSY; + } + } + mpu->port = port; + if (irq >= 0 && irq_flags) { + if (request_irq(irq, snd_msndmidi_interrupt, irq_flags, "MPU401 UART", (void *) mpu)) { + snd_printk("unable to grab IRQ %d\n", irq); + snd_device_free(card, rmidi); + return -EBUSY; + } + mpu->irq = irq; + mpu->irq_flags = irq_flags; + }*/ + strcpy(rmidi->name, "MSND MIDI"); + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_msndmidi_input); + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT ; + mpu->rmidi = rmidi; + return 0; +} diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c new file mode 100644 index 0000000..18b1e07 --- /dev/null +++ b/sound/isa/msnd/msnd_pinnacle.c @@ -0,0 +1,1894 @@ +/********************************************************************* + * + * Linux multisound pinnacle/fiji driver for alsa 0.9rc2CVS + * + * 2002/06/30 Karsten Wiese: + * for now this is only used to build a pinnacle / fiji driver. + * the OSS parent of this code is designed to also support + * the multisound classic via the file msnd_classic.c. + * to make it easier for some brave heart to implemt classic + * support in alsa, i left all the MSND_CLASSIC tokens in this file. + * but for now this untested & undone. + * + * + * ripped from linux kernel 2.4.18 by Karsten Wiese. + * + * the following is a copy of the 2.4.18 OSS FREE file-heading comment: + * + * Turtle Beach MultiSound Sound Card Driver for Linux + * msnd_pinnacle.c / msnd_classic.c + * + * -- If MSND_CLASSIC is defined: + * + * -> driver for Turtle Beach Classic/Monterey/Tahiti + * + * -- Else + * + * -> driver for Turtle Beach Pinnacle/Fiji + * + * 12-3-2000 Modified IO port validation Steve Sycamore + * + * Copyright (C) 1998 Andrew Veliath + * + * 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., 675 Mass Ave, Cambridge, MA 02139, USA. + * + ********************************************************************/ + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/asound.h> +#include <sound/pcm.h> +#include <sound/mpu401.h> +#include <linux/isa.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/slab.h> +#include <linux/types.h> +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/ioport.h> +#include <linux/firmware.h> +#include <linux/pnp.h> +#include <linux/irq.h> +#include <linux/io.h> + + +#ifdef MSND_CLASSIC +# ifndef __alpha__ +# define SLOWIO +# endif +#endif +#include "msnd.h" +#ifdef MSND_CLASSIC +# ifdef CONFIG_MSNDCLAS_HAVE_BOOT +# define HAVE_DSPCODEH +# endif +# include "msnd_classic.h" +# define LOGNAME "msnd_classic" +#else +# ifdef CONFIG_MSNDPIN_HAVE_BOOT +# define HAVE_DSPCODEH +# endif +# include "msnd_pinnacle.h" +# define LOGNAME "snd_msnd_pinnacle" +#endif + +/* functions from external mixer file */ +void snd_msndmix_setup(struct snd_msnd *chip); +unsigned long snd_msndmix_force_recsrc(struct snd_msnd *chip, int recsrc); +int __devinit snd_msndmix_new(struct snd_card *card); + +static inline long get_play_delay_jiffies(struct snd_msnd *chip, long size) +{ + long tmp = (size * HZ * chip->play_sample_size) / 8; + return tmp / (chip->play_sample_rate * chip->play_channels); +} + +static inline long get_rec_delay_jiffies(struct snd_msnd *chip, long size) +{ + long tmp = (size * HZ * chip->capture_sample_size) / 8; + return tmp / (chip->capture_sample_rate * chip->capture_channels); +} + +#ifndef HAVE_DSPCODEH +static const u8 *dspini, *permini; +static int sizeof_dspini, sizeof_permini; +#endif + +static int snd_msnd_dsp_full_reset(struct snd_card *card); + +int snd_msnd_send_dsp_cmd_chk(struct snd_msnd *chip, u8 cmd) +{ + if (snd_msnd_send_dsp_cmd(chip, cmd) == 0) + return 0; + snd_msnd_dsp_full_reset(chip->card); + return snd_msnd_send_dsp_cmd(chip, cmd); +} + +static void snd_msnd_play_reset_queue(struct snd_msnd *chip, + unsigned int pcm_periods, + unsigned int pcm_count) +{ + int n; + unsigned long lpDAQ; + + chip->last_playbank = -1; + chip->playLimit = pcm_count * (pcm_periods - 1); + chip->playPeriods = pcm_periods; + isa_writew(PCTODSP_OFFSET(0 * DAQDS__size), chip->DAPQ + JQS_wHead); + isa_writew(PCTODSP_OFFSET(0 * DAQDS__size), chip->DAPQ + JQS_wTail); + + chip->play_period_bytes = pcm_count; + + lpDAQ = chip->base + DAPQ_DATA_BUFF; + for (n = 0; n < 3; ++n, lpDAQ += DAQDS__size) { + isa_writew(PCTODSP_BASED((u32)(pcm_count * (n % pcm_periods))), lpDAQ + DAQDS_wStart); + isa_writew(0, lpDAQ + DAQDS_wSize); + isa_writew(1, lpDAQ + DAQDS_wFormat); + isa_writew(chip->play_sample_size, lpDAQ + DAQDS_wSampleSize); + isa_writew(chip->play_channels, lpDAQ + DAQDS_wChannels); + isa_writew(chip->play_sample_rate, lpDAQ + DAQDS_wSampleRate); + isa_writew(HIMT_PLAY_DONE * 0x100 + n, lpDAQ + DAQDS_wIntMsg); + isa_writew(n, lpDAQ + DAQDS_wFlags); + } +} + +static void snd_msnd_capture_reset_queue(struct snd_msnd *chip, + unsigned int pcm_periods, + unsigned int pcm_count) +{ + int n; + unsigned long lpDAQ; + //unsigned long flags; + +// snd_msnd_init_queue(chip->DARQ, DARQ_DATA_BUFF, DARQ_BUFF_SIZE); + + chip->last_recbank = 2; + chip->captureLimit = pcm_count * (pcm_periods - 1); + chip->capturePeriods = pcm_periods; + isa_writew(PCTODSP_OFFSET(0 * DAQDS__size), chip->DARQ + JQS_wHead); + isa_writew(PCTODSP_OFFSET(chip->last_recbank * DAQDS__size), chip->DARQ + JQS_wTail); + + /* Critical section: bank 1 access. this is how the OSS driver does it: + spin_lock_irqsave(&chip->lock, flags); + outb(HPBLKSEL_1, chip->io + HP_BLKS); + isa_memset_io(chip->base, 0, DAR_BUFF_SIZE * 3); + outb(HPBLKSEL_0, chip->io + HP_BLKS); + spin_unlock_irqrestore(&chip->lock, flags);*/ + + chip->capturePeriodBytes = pcm_count; + //snd_printd("snd_msnd_capture_reset_queue() %i\n", pcm_count); + + for (n = 0, lpDAQ = chip->base + DARQ_DATA_BUFF; n < 3/*pcm_periods*/; ++n, lpDAQ += DAQDS__size) { + isa_writew(PCTODSP_BASED((u32)(pcm_count * (n % pcm_periods)) + 0x3000), lpDAQ + DAQDS_wStart); + isa_writew(pcm_count, lpDAQ + DAQDS_wSize); + isa_writew(1, lpDAQ + DAQDS_wFormat); + isa_writew(chip->capture_sample_size, lpDAQ + DAQDS_wSampleSize); + isa_writew(chip->capture_channels, lpDAQ + DAQDS_wChannels); + isa_writew(chip->capture_sample_rate, lpDAQ + DAQDS_wSampleRate); + isa_writew(HIMT_RECORD_DONE * 0x100 + n, lpDAQ + DAQDS_wIntMsg); + isa_writew(n, lpDAQ + DAQDS_wFlags); + } +} + +#ifdef NO0 +static void reset_queues(struct snd_msnd *chip) +{ + if (chip->mode & FMODE_WRITE) { + msnd_fifo_make_empty(&(chip->DAPF)); + snd_msnd_reset_play_queue(); + } + if (chip->mode & FMODE_READ) { + msnd_fifo_make_empty(&(chip.DARF)); + snd_msnd_reset_capture_queue(); + } +} +#endif + + +static void dsp_write_flush(struct snd_msnd *chip) +{ + if (!(chip->mode & FMODE_WRITE) || !test_bit(F_WRITING, &chip->flags)) + return; + set_bit(F_WRITEFLUSH, &chip->flags); +/* interruptible_sleep_on_timeout( + &chip->writeflush, + get_play_delay_jiffies(&chip, chip->DAPF.len));*/ + clear_bit(F_WRITEFLUSH, &chip->flags); + if (!signal_pending(current)) + schedule_timeout_interruptible(get_play_delay_jiffies(chip, chip->play_period_bytes)); + clear_bit(F_WRITING, &chip->flags); +} + +static void dsp_halt(struct snd_msnd *chip, struct file *file) +{ + if ((file ? file->f_mode : chip->mode) & FMODE_READ) { + clear_bit(F_READING, &chip->flags); + snd_msnd_send_dsp_cmd_chk(chip, HDEX_RECORD_STOP); + snd_msnd_disable_irq(chip); + if (file) { + printk(KERN_DEBUG LOGNAME ": Stopping read for %p\n", file); + chip->mode &= ~FMODE_READ; + } + clear_bit(F_AUDIO_READ_INUSE, &chip->flags); + } + if ((file ? file->f_mode : chip->mode) & FMODE_WRITE) { + if (test_bit(F_WRITING, &chip->flags)) { + dsp_write_flush(chip); + snd_msnd_send_dsp_cmd_chk(chip, HDEX_PLAY_STOP); + } + snd_msnd_disable_irq(chip); + if (file) { + printk(KERN_DEBUG LOGNAME ": Stopping write for %p\n", file); + chip->mode &= ~FMODE_WRITE; + } + clear_bit(F_AUDIO_WRITE_INUSE, &chip->flags); + } +} + +#ifdef NO0 +static int dsp_release(struct file *file) +{ + dsp_halt(file); + return 0; +} +#endif + + +static void set_default_audio_parameters(struct snd_msnd *chip) +{ + chip->play_sample_size = DEFSAMPLESIZE; + chip->play_sample_rate = DEFSAMPLERATE; + chip->play_channels = DEFCHANNELS; + chip->capture_sample_size = DEFSAMPLESIZE; + chip->capture_sample_rate = DEFSAMPLERATE; + chip->capture_channels = DEFCHANNELS; +} + + +static int snd_msnd_DARQ(struct snd_msnd *chip, int bank) +{ + int /*size, n,*/ timeout = 3; + u16 wTmp; + //unsigned long DAQD; + + /* Increment the tail and check for queue wrap */ + wTmp = isa_readw(chip->DARQ + JQS_wTail) + PCTODSP_OFFSET(DAQDS__size); + //printk(KERN_DEBUG "%iR wTmp = %i", bank, (int)wTmp); + if (wTmp > isa_readw(chip->DARQ + JQS_wSize)) + wTmp = 0; + //printk(KERN_DEBUG " %i\n", (int)wTmp); + while (wTmp == isa_readw(chip->DARQ + JQS_wHead) && timeout--) + udelay(1); + + if (chip->capturePeriods == 2) { + unsigned long lpDAQ = chip->base + DARQ_DATA_BUFF + bank * DAQDS__size + DAQDS_wStart; + unsigned short offset = isa_readw(lpDAQ); + isa_writew(offset == PCTODSP_BASED(0x3000) ? PCTODSP_BASED(0x3000 + chip->capturePeriodBytes) : PCTODSP_BASED(0x3000), lpDAQ); + } + + isa_writew(wTmp, chip->DARQ + JQS_wTail); + + /* Get our digital audio queue struct + DAQD = bank * DAQDS__size + chip->base + DARQ_DATA_BUFF;*/ + + /* Get length of data */ + //size = isa_readw(DAQD + DAQDS_wSize); + + /* Read data from the head (unprotected bank 1 access okay + since this is only called inside an interrupt) */ +// outb(HPBLKSEL_1, chip->io + HP_BLKS); +/* if ((n = msnd_fifo_write( + &chip->DARF, + (char *)(chip->base + bank * DAR_BUFF_SIZE), + size, 0)) <= 0) { + outb(HPBLKSEL_0, chip->io + HP_BLKS); + return n; + }*/ +// outb(HPBLKSEL_0, chip->io + HP_BLKS); + + return 1; +} + +static int snd_msnd_DAPQ(struct snd_msnd *chip, int start) +{ + u16 DAPQ_tail; + int protect = start, nbanks = 0; + unsigned long DAQD; + static int play_banks_submitted; + //unsigned long flags; + //spin_lock_irqsave(&chip->lock, flags); not necessary + + DAPQ_tail = isa_readw(chip->DAPQ + JQS_wTail); + while (DAPQ_tail != isa_readw(chip->DAPQ + JQS_wHead) || start) { + int bank_num = DAPQ_tail / PCTODSP_OFFSET(DAQDS__size); + + if (start) { + start = 0; + play_banks_submitted = 0; + } + + /* Get our digital audio queue struct */ + DAQD = bank_num * DAQDS__size + chip->base + DAPQ_DATA_BUFF; + + /* Write size of this bank */ + isa_writew(chip->play_period_bytes, DAQD + DAQDS_wSize); + if (play_banks_submitted < 3) + ++play_banks_submitted; + else { + if (chip->playPeriods == 2) { + unsigned short offset = isa_readw(DAQD + DAQDS_wStart); + isa_writew(offset == PCTODSP_BASED(0x0) ? PCTODSP_BASED(0x0 + chip->play_period_bytes): PCTODSP_BASED(0x0), DAQD + DAQDS_wStart); + } + } + ++nbanks; + + /* Then advance the tail */ + /* + if (protect) + snd_printd("B %X %lX\n", bank_num, xtime.tv_usec); + */ + + DAPQ_tail = (++bank_num % 3) * PCTODSP_OFFSET(DAQDS__size); + isa_writew(DAPQ_tail, chip->DAPQ + JQS_wTail); + /* Tell the DSP to play the bank */ + snd_msnd_send_dsp_cmd(chip, HDEX_PLAY_START); + if (protect) + if (2 == bank_num) + break; + } + /* + if (protect) + snd_printd("%lX\n", xtime.tv_usec); + */ + //spin_unlock_irqrestore(&chip->lock, flags); not necessary + return nbanks; +} + + +static int InTrigger; // interrupt diagnostic, comment this out later +static int banksPlayed; +//static int playPosQueriesSinceInt; +//static int play_bytes_remaining_last; +//static int play_bytes_jiffies_last; + +static void snd_msnd_eval_dsp_msg(struct snd_msnd *chip, u16 wMessage) +{ + switch (HIBYTE(wMessage)) { + case HIMT_PLAY_DONE: { + //snd_printd("snd_msnd_eval_dsp_msg(%i)\n", wMessage); + #ifdef CONFIG_SND_DEBUG0 + if (banksPlayed < 3) + printk(KERN_DEBUG "%08X: HIMT_PLAY_DONE: %i\n", (unsigned)jiffies, LOBYTE(wMessage)); + #endif + #ifdef CONFIG_SND_DEBUG0 + { + int xx = *(short int*)(__ISA_IO_base + 0x7F40 + dev->base); + printk(KERN_DEBUG "%08X: P %X\n", (unsigned)jiffies, xx); + } + #endif + + if (chip->last_playbank == LOBYTE(wMessage)) { + snd_printd("chip.last_playbank == LOBYTE(wMessage)\n"); + break; + } + banksPlayed++; + + if (test_bit(F_WRITING, &chip->flags)) + snd_msnd_DAPQ(chip, 0); + + chip->last_playbank = LOBYTE(wMessage); + chip->playDMAPos += chip->play_period_bytes; + if (chip->playDMAPos > chip->playLimit) + chip->playDMAPos = 0; + //playPosQueriesSinceInt = 0; + snd_pcm_period_elapsed(chip->playback_substream); + //play_bytes_remaining_last += chip->play_period_bytes; + + break; + } + case HIMT_RECORD_DONE: + if (chip->last_recbank == LOBYTE(wMessage)) + break; + chip->last_recbank = LOBYTE(wMessage); + chip->captureDMAPos += chip->capturePeriodBytes; + if (chip->captureDMAPos > (chip->captureLimit)) + chip->captureDMAPos = 0; + + if (test_bit(F_READING, &chip->flags)) + snd_msnd_DARQ(chip, chip->last_recbank); + + snd_pcm_period_elapsed(chip->capture_substream); + break; + + case HIMT_DSP: + switch (LOBYTE(wMessage)) { +#ifndef MSND_CLASSIC + case HIDSP_PLAY_UNDER: +#endif + case HIDSP_INT_PLAY_UNDER: + printk(KERN_DEBUG LOGNAME ": Play underflow %i\n", banksPlayed); + if (banksPlayed > 2) + clear_bit(F_WRITING, &chip->flags); + break; + + case HIDSP_INT_RECORD_OVER: + printk(KERN_DEBUG LOGNAME ": Record overflow\n"); + clear_bit(F_READING, &chip->flags); + break; + + default: + printk(KERN_DEBUG LOGNAME ": DSP message %d 0x%02x\n", + LOBYTE(wMessage), LOBYTE(wMessage)); + break; + } + break; + + case HIMT_MIDI_IN_UCHAR: +printk(KERN_DEBUG "msnd midi int"); + if (chip->msndmidi_mpu) + snd_msndmidi_input_read(chip->msndmidi_mpu); + break; + + default: + printk(KERN_DEBUG LOGNAME ": HIMT message %d 0x%02x\n", HIBYTE(wMessage), HIBYTE(wMessage)); + break; + } +} + +//static int InInterrupt = 0; +irqreturn_t snd_msnd_interrupt(int irq, void *dev_id) +{ + struct snd_msnd *chip = dev_id; + /*if (InInterrupt) { + printk(KERN_DEBUG "INTERRUPT in InInterrupt\n"); + return IRQ_NONE; + } + InInterrupt = 1;*/ +#ifdef CONFIG_SND_DEBUG + // interrupt diagnostic, comment this out later + if (InTrigger) + printk(KERN_DEBUG "INTERRUPT in InTrigger %i\n", InTrigger); // should never happen +#endif + /* Send ack to DSP */ +// inb(chip->io + HP_RXL); + + /* Evaluate queued DSP messages */ + while (isa_readw(chip->DSPQ + JQS_wTail) != isa_readw(chip->DSPQ + JQS_wHead)) { + u16 wTmp; + + snd_msnd_eval_dsp_msg(chip, isa_readw(chip->pwDSPQData + 2*isa_readw(chip->DSPQ + JQS_wHead))); + + wTmp = isa_readw(chip->DSPQ + JQS_wHead) + 1; + if (wTmp > isa_readw(chip->DSPQ + JQS_wSize)) + isa_writew(0, chip->DSPQ + JQS_wHead); + else + isa_writew(wTmp, chip->DSPQ + JQS_wHead); + } + /* Send ack to DSP */ + inb(chip->io + HP_RXL); + //InInterrupt = 0; + return IRQ_HANDLED; +} + + +static int snd_msnd_reset_dsp(long io, unsigned char *info) +{ + int timeout = 100; + + outb(HPDSPRESET_ON, io + HP_DSPR); + msleep(1); +#ifndef MSND_CLASSIC + if (info) + *info = inb(io + HP_INFO); +#endif + outb(HPDSPRESET_OFF, io + HP_DSPR); + msleep(1); + while (timeout-- > 0) { + if (inb(io + HP_CVR) == HP_CVR_DEF) + return 0; + msleep(1); + } + printk(KERN_ERR LOGNAME ": Cannot reset DSP\n"); + + return -EIO; +} + +static int __devinit snd_msnd_probe(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + unsigned char info; +#ifndef MSND_CLASSIC + char *xv, *rev = NULL; + char *pin = "TB Pinnacle", *fiji = "TB Fiji"; + char *pinfiji = "TB Pinnacle/Fiji"; +#endif + + if (!request_region(chip->io, DSP_NUMIO, "probing")) { + printk(KERN_ERR LOGNAME ": I/O port conflict\n"); + return -ENODEV; + } + + if (snd_msnd_reset_dsp(chip->io, &info) < 0) { + release_region(chip->io, DSP_NUMIO); + return -ENODEV; + } + +#ifdef MSND_CLASSIC + strcpy(card->shortname, "Classic/Tahiti/Monterey"); + printk(KERN_INFO LOGNAME ": %s, " +#else + switch (info >> 4) { + case 0xf: + xv = "<= 1.15"; + break; + case 0x1: + xv = "1.18/1.2"; + break; + case 0x2: + xv = "1.3"; + break; + case 0x3: + xv = "1.4"; + break; + default: + xv = "unknown"; + break; + } + + switch (info & 0x7) { + case 0x0: + rev = "I"; + strcpy(card->shortname, pin); + break; + case 0x1: + rev = "F"; + strcpy(card->shortname, pin); + break; + case 0x2: + rev = "G"; + strcpy(card->shortname, pin); + break; + case 0x3: + rev = "H"; + strcpy(card->shortname, pin); + break; + case 0x4: + rev = "E"; + strcpy(card->shortname, fiji); + break; + case 0x5: + rev = "C"; + strcpy(card->shortname, fiji); + break; + case 0x6: + rev = "D"; + strcpy(card->shortname, fiji); + break; + case 0x7: + rev = "A-B (Fiji) or A-E (Pinnacle)"; + strcpy(card->shortname, pinfiji); + break; + } + printk(KERN_INFO LOGNAME ": %s revision %s, Xilinx version %s, " +#endif /* MSND_CLASSIC */ + "I/O 0x%lx-0x%lx, IRQ %d, memory mapped to 0x%lX-0x%lX\n", + card->shortname, +#ifndef MSND_CLASSIC + rev, xv, +#endif + chip->io, chip->io + DSP_NUMIO - 1, + chip->irq, + chip->base, chip->base + 0x7fff); + + strcpy(card->longname, "Turtle Beach Multisound Pinnacle"); + release_region(chip->io, DSP_NUMIO); + return 0; +} + +static int snd_msnd_init_sma(struct snd_msnd *chip) +{ + static int initted; + u16 mastVolLeft, mastVolRight; + unsigned long flags; + +#ifdef MSND_CLASSIC + outb(chip->memid, chip->io + HP_MEMM); +#endif + outb(HPBLKSEL_0, chip->io + HP_BLKS); + if (initted) { + mastVolLeft = isa_readw(chip->SMA + SMA_wCurrMastVolLeft); + mastVolRight = isa_readw(chip->SMA + SMA_wCurrMastVolRight); + } else + mastVolLeft = mastVolRight = 0; + isa_memset_io(chip->base, 0, 0x8000); + + /* Critical section: bank 1 access */ + spin_lock_irqsave(&chip->lock, flags); + outb(HPBLKSEL_1, chip->io + HP_BLKS); + isa_memset_io(chip->base, 0, 0x8000); + outb(HPBLKSEL_0, chip->io + HP_BLKS); + spin_unlock_irqrestore(&chip->lock, flags); + + chip->pwDSPQData = (chip->base + DSPQ_DATA_BUFF); + chip->pwMODQData = (chip->base + MODQ_DATA_BUFF); + chip->pwMIDQData = (chip->base + MIDQ_DATA_BUFF); + + /* Motorola 56k shared memory base */ + chip->SMA = chip->base + SMA_STRUCT_START; + + /* Digital audio play queue */ + chip->DAPQ = chip->base + DAPQ_OFFSET; + snd_msnd_init_queue(chip->DAPQ, DAPQ_DATA_BUFF, DAPQ_BUFF_SIZE); + + /* Digital audio record queue */ + chip->DARQ = chip->base + DARQ_OFFSET; + snd_msnd_init_queue(chip->DARQ, DARQ_DATA_BUFF, DARQ_BUFF_SIZE); + + /* MIDI out queue */ + chip->MODQ = chip->base + MODQ_OFFSET; + snd_msnd_init_queue(chip->MODQ, MODQ_DATA_BUFF, MODQ_BUFF_SIZE); + + /* MIDI in queue */ + chip->MIDQ = chip->base + MIDQ_OFFSET; + snd_msnd_init_queue(chip->MIDQ, MIDQ_DATA_BUFF, MIDQ_BUFF_SIZE); + + /* DSP -> host message queue */ + chip->DSPQ = chip->base + DSPQ_OFFSET; + snd_msnd_init_queue(chip->DSPQ, DSPQ_DATA_BUFF, DSPQ_BUFF_SIZE); + + /* Setup some DSP values */ +#ifndef MSND_CLASSIC + isa_writew(1, chip->SMA + SMA_wCurrPlayFormat); + isa_writew(chip->play_sample_size, chip->SMA + SMA_wCurrPlaySampleSize); + isa_writew(chip->play_channels, chip->SMA + SMA_wCurrPlayChannels); + isa_writew(chip->play_sample_rate, chip->SMA + SMA_wCurrPlaySampleRate); +#endif + isa_writew(chip->play_sample_rate, chip->SMA + SMA_wCalFreqAtoD); + isa_writew(mastVolLeft, chip->SMA + SMA_wCurrMastVolLeft); + isa_writew(mastVolRight, chip->SMA + SMA_wCurrMastVolRight); +#ifndef MSND_CLASSIC + isa_writel(0x00010000, chip->SMA + SMA_dwCurrPlayPitch); + isa_writel(0x00000001, chip->SMA + SMA_dwCurrPlayRate); +#endif + isa_writew(0x303, chip->SMA + SMA_wCurrInputTagBits); + + initted = 1; + + return 0; +} + + +//////////////////////////////////////////////////////////////////////////////// + +static int __devinit snd_msnd_calibrate_adc(struct snd_msnd *chip, u16 srate) +{ + snd_printd("snd_msnd_calibrate_adc(%i)\n", srate); + isa_writew(srate, chip->SMA + SMA_wCalFreqAtoD); + if (chip->calibrate_signal == 0) + isa_writew(isa_readw(chip->SMA + SMA_wCurrHostStatusFlags) + | 0x0001, chip->SMA + SMA_wCurrHostStatusFlags); + else + isa_writew(isa_readw(chip->SMA + SMA_wCurrHostStatusFlags) + & ~0x0001, chip->SMA + SMA_wCurrHostStatusFlags); + if (snd_msnd_send_word(chip, 0, 0, HDEXAR_CAL_A_TO_D) == 0 && + snd_msnd_send_dsp_cmd_chk(chip, HDEX_AUX_REQ) == 0) { + schedule_timeout_interruptible(msecs_to_jiffies(333)); + return 0; + } + printk(KERN_WARNING LOGNAME ": ADC calibration failed\n"); + return -EIO; +} + +static int upload_dsp_code(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; +#ifndef HAVE_DSPCODEH + const struct firmware *init_fw = NULL, *perm_fw = NULL; +#endif + int err; + + outb(HPBLKSEL_0, chip->io + HP_BLKS); +#ifndef HAVE_DSPCODEH + + err = request_firmware(&init_fw, INITCODEFILE, card->dev); + if (err < 0) { + printk(KERN_ERR LOGNAME ": Error loading " INITCODEFILE); + goto cleanup1; + } + err = request_firmware(&perm_fw, PERMCODEFILE, card->dev); + if (err < 0) { + printk(KERN_ERR LOGNAME ": Error loading " PERMCODEFILE); + goto cleanup; + } + INITCODE = init_fw->data; + INITCODESIZE = init_fw->size; + PERMCODE = perm_fw->data; + PERMCODESIZE = perm_fw->size; +#endif + isa_memcpy_toio(chip->base, PERMCODE, PERMCODESIZE); + if (snd_msnd_upload_host(chip, INITCODE, INITCODESIZE) < 0) { + printk(KERN_WARNING LOGNAME ": Error uploading to DSP\n"); + err = -ENODEV; + goto cleanup; + } +#ifdef HAVE_DSPsleep + printk(KERN_INFO LOGNAME ": DSP firmware uploaded (resident)\n"); +#else + printk(KERN_INFO LOGNAME ": DSP firmware uploaded\n"); +#endif + err = 0; + +cleanup: +#ifndef HAVE_DSPCODEH + release_firmware(perm_fw); +cleanup1: + release_firmware(init_fw); +#endif + return err; +} + +#ifdef MSND_CLASSIC +static void reset_proteus(struct snd_msnd *chip) +{ + outb(HPPRORESET_ON, chip->io + HP_PROR); + msleep(TIME_PRO_RESET); + outb(HPPRORESET_OFF, chip->io + HP_PROR); + msleep(TIME_PRO_RESET_DONE); +} +#endif + +static int snd_msnd_initialize(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + int err, timeout; + //snd_printd("snd_msnd_initialize(void)\n"); + +#ifdef MSND_CLASSIC + outb(HPWAITSTATE_0, chip->io + HP_WAIT); + outb(HPBITMODE_16, chip->io + HP_BITM); + + reset_proteus(chip); +#endif + err = snd_msnd_init_sma(chip); + if (err < 0) { + printk(KERN_WARNING LOGNAME ": Cannot initialize SMA\n"); + return err; + } + + err = snd_msnd_reset_dsp(chip->io, NULL); + if (err < 0) + return err; + + err = upload_dsp_code(card); + if (err < 0) { + printk(KERN_WARNING LOGNAME ": Cannot upload DSP code\n"); + return err; + } + + timeout = 200; + + //snd_printd("%li\n", chip->base); + while (isa_readw(chip->base)) { + msleep(1); + if (!timeout--) { + printk(KERN_DEBUG LOGNAME ": DSP reset timeout\n"); + return -EIO; + } + } + + snd_msndmix_setup(chip); + return 0; +} + +static int snd_msnd_dsp_full_reset(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + int rv; + + if (test_bit(F_RESETTING, &chip->flags) || ++chip->nresets > 10) + return 0; + + set_bit(F_RESETTING, &chip->flags); + dsp_halt(chip, NULL); /* Unconditionally halt */ + + rv = snd_msnd_initialize(card); + if (rv) + printk(KERN_WARNING LOGNAME ": DSP reset failed\n"); + snd_msndmix_force_recsrc(chip, 0); + clear_bit(F_RESETTING, &chip->flags); + return rv; +} + +static int snd_msnd_dev_free(struct snd_device *device) +{ + snd_printd("snd_msnd_chip_free()\n"); + return 0; +} + +static struct snd_pcm_hardware snd_msnd_playback = { + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP_VALID), + .formats = (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE), + .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = 0x3000, + .period_bytes_min = 0x40, + .period_bytes_max = 0x1800, + .periods_min = 2, + .periods_max = 3, + .fifo_size = 0, +}; + +static struct snd_pcm_hardware snd_msnd_capture = { + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE), + .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = 0x3000, + .period_bytes_min = 0x40, + .period_bytes_max = 0x1800, + .periods_min = 2, + .periods_max = 3, + .fifo_size = 0, +}; + + +static unsigned int rates[] = { + 8000, 11025, 16000, 22050, + 32000, 44100, 48000 +}; + +static struct snd_pcm_hw_constraint_list hw_constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static int snd_msnd_playback_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + //snd_printd("snd_msnd_playback_open()\n"); + + set_bit(F_AUDIO_WRITE_INUSE, &chip->flags); + clear_bit(F_WRITING, &chip->flags); + snd_msnd_enable_irq(chip); + + runtime->dma_area = chip->mappedbase; + //memset(__ISA_IO_base + chip->base, 0, 3*0x2400); + runtime->dma_bytes = 0x3000; + + chip->playback_substream = substream; + runtime->hw = snd_msnd_playback; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); + return 0; +} + +static int snd_msnd_playback_close(struct snd_pcm_substream *substream) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + //snd_printd("snd_msnd_playback_close()\n"); + snd_msnd_disable_irq(chip); + clear_bit(F_AUDIO_WRITE_INUSE, &chip->flags); + return 0; +} + + +static int snd_msnd_playback_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) +{ + int i; + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + unsigned long lpDAQ = chip->base + DAPQ_DATA_BUFF; + + //this results in 8 with the 2 synts amsynth & bristol so for now i hardcode it: + //chip->play_sample_size = snd_pcm_format_width(substream->runtime->format); + //printk(KERN_DEBUG "chip->play_sample_size %i\n", chip->play_sample_size); + chip->play_sample_size = 16; + chip->play_channels = params_channels(params); + chip->play_sample_rate = params_rate(params); + + //snd_printd("snd_msnd_playback_hw_params()\n"); + //snd_printd("f: %i; c: %i; r: %i\n", chip->play_sample_size, chip->play_channels, chip->play_sample_rate); + + for (i = 0; i < 3; ++i, lpDAQ += DAQDS__size) { + isa_writew(chip->play_sample_size, lpDAQ + DAQDS_wSampleSize); + isa_writew(chip->play_channels, lpDAQ + DAQDS_wChannels); + isa_writew(chip->play_sample_rate, lpDAQ + DAQDS_wSampleRate); + } + // dont do this here: snd_msnd_calibrate_adc(chip->play_sample_rate); + + return 0; +} + + + +/*static int snd_msnd_playback_hw_free(struct snd_pcm_substream *substream) +{ + snd_printd("snd_msnd_playback_hw_free()\n"); + return 0; +} */ + +static int snd_msnd_playback_prepare(struct snd_pcm_substream *substream) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + unsigned int pcm_size = snd_pcm_lib_buffer_bytes(substream); + unsigned int pcm_count = snd_pcm_lib_period_bytes(substream); + unsigned int pcm_periods = pcm_size / pcm_count; + + //snd_printd("snd_msnd_playback_prepare()\n"); + //snd_printd("buffer_bytes=%i; period_bytes=%i\n", pcm_size, pcm_count); + snd_msnd_play_reset_queue(chip, pcm_periods, pcm_count); + chip->playDMAPos = 0; + return 0; +} + +//static int played_bytes; +static int snd_msnd_playback_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + int result = 0; + + //spin_lock(&chip->reg_lock); + if (cmd == SNDRV_PCM_TRIGGER_START) { + //snd_printd("snd_msnd_playback_trigger(START)\n"); + InTrigger = 1; // interrupt diagnostic, comment this out later + //play_bytes_remaining_last = 0; + banksPlayed = 0; + //playPosQueriesSinceInt = 0; + //played_bytes = -1; + set_bit(F_WRITING, &chip->flags); + // this gives looong timeouts, so dont do it here: snd_msnd_calibrate_adc(chip->play_sample_rate); + snd_msnd_DAPQ(chip, 1); + } else if (cmd == SNDRV_PCM_TRIGGER_STOP) { + //snd_printd("snd_msnd_playback_trigger(STop)\n"); + InTrigger = 2; // interrupt diagnostic, comment this out later + clear_bit(F_WRITING, &chip->flags); + snd_msnd_send_dsp_cmd(chip, HDEX_PLAY_STOP); + } else { + snd_printd("snd_msnd_playback_trigger(?????)\n"); + result = -EINVAL; + } + InTrigger = 0; // interrupt diagnostic, comment this out later + // spin_unlock(&chip->reg_lock); + + //snd_printd("snd_msnd_playback_trigger() ENDE\n"); + return result; +} + +static snd_pcm_uframes_t snd_msnd_playback_pointer(struct snd_pcm_substream *substream) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + //snd_printd("snd_msnd_playback_pointer()\n"); + + /* with the following mess i tried to generate a more precise pointer position + * it generated errors with the alsa framework i could not resolve..... + //int pos = dev.playDMAPos; + unsigned remaining = *(short int*)(__ISA_IO_base + 0x7F40 + dev.base), + ljiffies = (unsigned)jiffies; + int diff; + if (-1 == played_bytes) { + if (0 == remaining) + played_bytes = 0; + else + played_bytes = dev.play_period_bytes - remaining; + // played_bytes += 0x60; + } else { + int ref = (ljiffies - play_bytes_jiffies_last) + * (runtime->frame_bits / 16) + * (runtime->rate /HZ); + diff = play_bytes_remaining_last - remaining; + //printk(KERN_DEBUG "pb %i diff %i ref %i", played_bytes, diff, ref); + if (diff < ref) { + diff += dev.play_period_bytes; + //printk(" newdiff %i ", diff); + } + played_bytes += diff; + //printk("\n"); + } + + play_bytes_remaining_last = remaining; + play_bytes_jiffies_last = ljiffies; + played_bytes %= snd_msnd_playback.buffer_bytes_max; + + // if (playPosQueriesSinceInt++) + // pos += dev.play_period_bytes - remaining; + + { + snd_printd("%08X: remaining %04X\n", (unsigned)jiffies, remaining); + snd_printd("snd_msnd_playback_pointer() %X\n", played_bytes); + } + + diff = dev.playDMAPos - played_bytes; + if (diff < 0) + diff += snd_msndpinnacle_playback.buffer_bytes_max; + + return bytes_to_frames(runtime, + diff < (snd_msndpinnacle_playback.buffer_bytes_max / 2) + ? dev.playDMAPos : played_bytes); + // end of mess + */ + return bytes_to_frames(substream->runtime, chip->playDMAPos); +} + + +static struct snd_pcm_ops snd_msnd_playback_ops = { + .open = snd_msnd_playback_open, + .close = snd_msnd_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_msnd_playback_hw_params, + .prepare = snd_msnd_playback_prepare, + .trigger = snd_msnd_playback_trigger, + .pointer = snd_msnd_playback_pointer, +}; + +static int snd_msnd_capture_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + set_bit(F_AUDIO_READ_INUSE, &chip->flags); + snd_msnd_enable_irq(chip); + runtime->dma_area = chip->mappedbase + 0x3000; + runtime->dma_bytes = 0x3000; + memset(runtime->dma_area, 0, runtime->dma_bytes); + chip->capture_substream = substream; + runtime->hw = snd_msnd_capture; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); + return 0; +} + +static int snd_msnd_capture_close(struct snd_pcm_substream *substream) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + snd_msnd_disable_irq(chip); + clear_bit(F_AUDIO_READ_INUSE, &chip->flags); + return 0; +} + +static int snd_msnd_capture_prepare(struct snd_pcm_substream *substream) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + unsigned int pcm_size = snd_pcm_lib_buffer_bytes(substream); + unsigned int pcm_count = snd_pcm_lib_period_bytes(substream); + unsigned int pcm_periods = pcm_size / pcm_count; + + snd_msnd_capture_reset_queue(chip, pcm_periods, pcm_count); + chip->captureDMAPos = 0; + return 0; +} + +static int snd_msnd_capture_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + if (cmd == SNDRV_PCM_TRIGGER_START) { + chip->last_recbank = -1; + set_bit(F_READING, &chip->flags); + if (snd_msnd_send_dsp_cmd_chk(chip, HDEX_RECORD_START) == 0) + return 0; + + clear_bit(F_READING, &chip->flags); + } else if (cmd == SNDRV_PCM_TRIGGER_STOP) { + clear_bit(F_READING, &chip->flags); + snd_msnd_send_dsp_cmd(chip, HDEX_RECORD_STOP); + return 0; + } + return -EINVAL; +} + + +static snd_pcm_uframes_t snd_msnd_capture_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + return bytes_to_frames(runtime, chip->captureDMAPos); +} + + +static int snd_msnd_capture_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) +{ + int i; + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + unsigned long lpDAQ = chip->base + DARQ_DATA_BUFF; + + //this results in 8 with the 2 synts amsynth & bristol so for now i hardcode it: + //chip->play_sample_size = snd_pcm_format_width(substream->runtime->format); + //printk(KERN_DEBUG "chip->play_sample_size %i\n", chip->play_sample_size); + chip->capture_sample_size = 16; + chip->capture_channels = params_channels(params); + chip->capture_sample_rate = params_rate(params); + + //snd_printd("snd_msnd_capture_hw_params()\n"); + //snd_printd("f: %i; c: %i; r: %i\n", chip->capture_sample_size, chip->capture_channels, chip->capture_sample_rate); + + for (i = 0; i < 3; ++i, lpDAQ += DAQDS__size) { + isa_writew(chip->capture_sample_size, lpDAQ + DAQDS_wSampleSize); + isa_writew(chip->capture_channels, lpDAQ + DAQDS_wChannels); + isa_writew(chip->capture_sample_rate, lpDAQ + DAQDS_wSampleRate); + } + return 0; +} + + +static struct snd_pcm_ops snd_msnd_capture_ops = { + .open = snd_msnd_capture_open, + .close = snd_msnd_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_msnd_capture_hw_params, + .prepare = snd_msnd_capture_prepare, + .trigger = snd_msnd_capture_trigger, + .pointer = snd_msnd_capture_pointer, +}; + + +int snd_msnd_pcm(struct snd_card *card, int device, struct snd_pcm **rpcm) +{ + struct snd_msnd *chip = card->private_data; + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(card, "MSNDPINNACLE", device, 1, 1, &pcm); + if (err < 0) + return err; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_msnd_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_msnd_capture_ops); + + pcm->private_free = NULL;//snd_msnd_pcm_free; + pcm->private_data = chip; + pcm->info_flags = 0;//SNDRV_PCM_INFO_HALF_DUPLEX; + strcpy(pcm->name, "Hurricane"); + + + if (rpcm) + *rpcm = pcm; + return 0; +} + +/* + * ALSA callback function, called when attempting to open the MIDI device. + */ +static int snd_msnd_mpu401_open(struct snd_mpu401 *mpu) +{ + snd_msnd_send_dsp_cmd(mpu->private_data, HDEX_MIDI_IN_START); + return 0; +} + +static void snd_msnd_mpu401_close(struct snd_mpu401 *mpu) +{ + snd_msnd_send_dsp_cmd(mpu->private_data, HDEX_MIDI_IN_STOP); +} + +static long mpu_io[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT; +static int mpu_irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ; + +static int __devinit snd_msnd_attach(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + int err; + static struct snd_device_ops ops = { + .dev_free = snd_msnd_dev_free, + }; + + err = request_irq(chip->irq, snd_msnd_interrupt, 0, card->shortname, + chip); + if (err < 0) { + printk(KERN_ERR LOGNAME ": Couldn't grab IRQ %d\n", chip->irq); + return err; + } + request_region(chip->io, DSP_NUMIO, card->shortname); + + if (!request_mem_region((unsigned)(__ISA_IO_base + chip->base), BUFFSIZE, card->shortname)) { + printk(KERN_ERR LOGNAME ": unable to grab memory region 0x%lx-0x%lx\n", + chip->base, chip->base + BUFFSIZE - 1); + release_region(chip->io, DSP_NUMIO); + free_irq(chip->irq, chip); + return -EBUSY; + } + chip->mappedbase = __ISA_IO_base + chip->base; + + snd_printd("chip->mappedbase = 0x%08X\n", (unsigned)chip->mappedbase); + + err = snd_msnd_dsp_full_reset(card); + if (err < 0) + goto err_release_region; + + /* Register device */ + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) + goto err_release_region; + + err = snd_msnd_pcm(card, 0, NULL); + if (err < 0) { + printk(KERN_ERR LOGNAME ": error creating new PCM device\n"); + goto err_release_region; + } + + err = snd_msndmix_new(card); + if (err < 0) { + printk(KERN_ERR LOGNAME ": error creating new Mixer device\n"); + goto err_release_region; + } + + + if (mpu_io[0] != SNDRV_AUTO_PORT) + { + struct snd_mpu401 *mpu; + + err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, + mpu_io[0], + MPU401_MODE_INPUT | + MPU401_MODE_OUTPUT, + mpu_irq[0], IRQF_DISABLED, + &chip->rmidi); + if (err < 0) { + printk(KERN_ERR LOGNAME ": error creating new Midi device\n"); + goto err_release_region; + } + mpu = chip->rmidi->private_data; + + mpu->open_input = snd_msnd_mpu401_open; + mpu->close_input = snd_msnd_mpu401_close; + mpu->private_data = chip; + } + + disable_irq(chip->irq); + snd_msnd_calibrate_adc(chip, chip->play_sample_rate); + snd_msndmix_force_recsrc(chip, 0); + + err = snd_card_register(card); + if (err < 0) + goto err_release_region; + + return 0; + +err_release_region: + release_mem_region(__ISA_IO_base + chip->base, BUFFSIZE); + release_region(chip->io, DSP_NUMIO); + free_irq(chip->irq, chip); + return err; +} + + +static void __devexit snd_msnd_unload(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + + release_mem_region(__ISA_IO_base + chip->base, BUFFSIZE); + release_region(chip->io, DSP_NUMIO); + free_irq(chip->irq, chip); + snd_card_free(card); +} + +#ifndef MSND_CLASSIC + +/* Pinnacle/Fiji Logical Device Configuration */ + +static int __devinit snd_msnd_write_cfg(int cfg, int reg, int value) +{ + outb(reg, cfg); + outb(value, cfg + 1); + if (value != inb(cfg + 1)) { + printk(KERN_ERR LOGNAME ": snd_msnd_write_cfg: I/O error\n"); + return -EIO; + } + return 0; +} + +static int __devinit snd_msnd_write_cfg_io0(int cfg, int num, u16 io) +{ + if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_IO0_BASEHI, HIBYTE(io))) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_IO0_BASELO, LOBYTE(io))) + return -EIO; + return 0; +} + +static int __devinit snd_msnd_write_cfg_io1(int cfg, int num, u16 io) +{ + if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_IO1_BASEHI, HIBYTE(io))) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_IO1_BASELO, LOBYTE(io))) + return -EIO; + return 0; +} + +static int __devinit snd_msnd_write_cfg_irq(int cfg, int num, u16 irq) +{ + if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_IRQ_NUMBER, LOBYTE(irq))) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_IRQ_TYPE, IRQTYPE_EDGE)) + return -EIO; + return 0; +} + +static int __devinit snd_msnd_write_cfg_mem(int cfg, int num, int mem) +{ + u16 wmem; + + mem >>= 8; + wmem = (u16)(mem & 0xfff); + if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_MEMBASEHI, HIBYTE(wmem))) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_MEMBASELO, LOBYTE(wmem))) + return -EIO; + if (wmem && snd_msnd_write_cfg(cfg, IREG_MEMCONTROL, (MEMTYPE_HIADDR | MEMTYPE_16BIT))) + return -EIO; + return 0; +} + +static int __devinit snd_msnd_activate_logical(int cfg, int num) +{ + if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_ACTIVATE, LD_ACTIVATE)) + return -EIO; + return 0; +} + +static int __devinit snd_msnd_write_cfg_logical(int cfg, int num, u16 io0, u16 io1, u16 irq, int mem) +{ + if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (snd_msnd_write_cfg_io0(cfg, num, io0)) + return -EIO; + if (snd_msnd_write_cfg_io1(cfg, num, io1)) + return -EIO; + if (snd_msnd_write_cfg_irq(cfg, num, irq)) + return -EIO; + if (snd_msnd_write_cfg_mem(cfg, num, mem)) + return -EIO; + if (snd_msnd_activate_logical(cfg, num)) + return -EIO; + return 0; +} + +static int __devinit snd_msnd_pinnacle_cfg_reset(int cfg) +{ + int i; + + /* Reset devices if told to */ + printk(KERN_INFO LOGNAME ": Resetting all devices\n"); + for (i = 0; i < 4; ++i) + if (snd_msnd_write_cfg_logical(cfg, i, 0, 0, 0, 0)) + return -EIO; + + return 0; +} +#endif + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ + +module_param_array(index, int, NULL, S_IRUGO); +MODULE_PARM_DESC(index, "Index value for msnd_pinnacle soundcard."); +module_param_array(id, charp, NULL, S_IRUGO); +MODULE_PARM_DESC(id, "ID string for msnd_pinnacle soundcard."); + +static long io[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT; +static int irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ; +static long mem[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT; + +static long cfg[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT; + +#ifndef MSND_CLASSIC +/* Extra Peripheral Configuration (Default: Disable) */ +static long ide_io0[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT; +static long ide_io1[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT; +static int ide_irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ; + +static long joystick_io[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT; +/* If we have the digital daugherboard... */ +static int digital[SNDRV_CARDS] __devinitdata; + +/* Extra Peripheral Configuration */ +static int reset[SNDRV_CARDS] __devinitdata; +#endif + +static int write_ndelay[SNDRV_CARDS] __devinitdata = + { [0 ... (SNDRV_CARDS-1)] = 1 }; + +#ifdef CONFIG_PNP +static int isapnp[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_ENABLE_PNP; +#endif + +#ifdef MODULE +MODULE_AUTHOR("Karsten Wiese <annabellesgarden(a)yahoo.de>"); +MODULE_DESCRIPTION("Turtle Beach " LONGNAME " Linux Driver"); +MODULE_LICENSE("GPL"); +#ifndef HAVE_DSPCODEH +MODULE_FIRMWARE(INITCODEFILE); +MODULE_FIRMWARE(PERMCODEFILE); +#endif + +#ifndef MSND_CLASSIC +//static int play_period_bytes __initdata = DAP_BUFF_SIZE; +#endif + +static int calibrate_signal __devinitdata; + +module_param_array(io, long, NULL, S_IRUGO); +MODULE_PARM_DESC(io, "IO port #"); +module_param_array(irq, int, NULL, S_IRUGO); +module_param_array(mem, long, NULL, S_IRUGO); +module_param_array(write_ndelay, int, NULL, S_IRUGO); +module_param(calibrate_signal, int, S_IRUGO); +#ifndef MSND_CLASSIC +module_param_array(digital, int, NULL, S_IRUGO); +module_param_array(cfg, long, NULL, S_IRUGO); +module_param_array(reset, int, 0, S_IRUGO); +module_param_array(mpu_io, long, NULL, S_IRUGO); +module_param_array(mpu_irq, int, NULL, S_IRUGO); +module_param_array(ide_io0, long, NULL, S_IRUGO); +module_param_array(ide_io1, long, NULL, S_IRUGO); +module_param_array(ide_irq, int, NULL, S_IRUGO); +module_param_array(joystick_io, long, NULL, S_IRUGO); +#ifdef CONFIG_PNP +module_param_array(isapnp, bool, NULL, 0444); +MODULE_PARM_DESC(isapnp, "ISA PnP detection for specified soundcard."); +#endif +#endif + +#else /* not a module */ + +#ifndef CONFIG_MSND_CALSIGNAL +# define CONFIG_MSND_CALSIGNAL 0 +#endif +static int +calibrate_signal __devinitdata = CONFIG_MSND_CALSIGNAL; +#endif /* MODULE */ + + +static int __devinit snd_msnd_isa_match(struct device *pdev, unsigned int i) +{ + if (io[i] == SNDRV_AUTO_PORT) + return 0; + + if (irq[i] == SNDRV_AUTO_PORT || mem[i] == SNDRV_AUTO_PORT) { + printk(KERN_WARNING LOGNAME ": io, irq and mem must be set\n"); + return 0; + } + +#ifdef MSND_CLASSIC + if (!(io[i] == 0x290 || + io[i] == 0x260 || + io[i] == 0x250 || + io[i] == 0x240 || + io[i] == 0x230 || + io[i] == 0x220 || + io[i] == 0x210 || + io[i] == 0x3e0)) { + printk(KERN_ERR LOGNAME ": \"io\" - DSP I/O base must be set to 0x210, 0x220, 0x230, 0x240, 0x250, 0x260, 0x290, or 0x3E0\n"); + return 0; + } +#else + if (io[i] < 0x100 || io[i] > 0x3e0 || (io[i] % 0x10) != 0) { + printk(KERN_ERR LOGNAME ": \"io\" - DSP I/O base must within the range 0x100 to 0x3E0 and must be evenly divisible by 0x10\n"); + return 0; + } +#endif /* MSND_CLASSIC */ + + if (!(irq[i] == 5 || + irq[i] == 7 || + irq[i] == 9 || + irq[i] == 10 || + irq[i] == 11 || + irq[i] == 12)) { + printk(KERN_ERR LOGNAME ": \"irq\" - must be set to 5, 7, 9, 10, 11 or 12\n"); + return 0; + } + + if (!(mem[i] == 0xb0000 || + mem[i] == 0xc8000 || + mem[i] == 0xd0000 || + mem[i] == 0xd8000 || + mem[i] == 0xe0000 || + mem[i] == 0xe8000)) { + printk(KERN_ERR LOGNAME ": \"mem\" - must be set to " + "0xb0000, 0xc8000, 0xd0000, 0xd8000, 0xe0000 or 0xe8000\n"); + return 0; + } + +#ifndef MSND_CLASSIC + if (cfg[i] == SNDRV_AUTO_PORT) { + printk(KERN_INFO LOGNAME ": Assuming PnP mode\n"); + } else if (cfg[i] != 0x250 && cfg[i] != 0x260 && cfg[i] != 0x270) { + printk(KERN_INFO LOGNAME ": Config port must be 0x250, 0x260 or 0x270 (or unspecified for PnP mode)\n"); + return 0; + } +#endif /* MSND_CLASSIC */ + + return 1; +} + +static int __devinit snd_msnd_isa_probe(struct device *pdev, unsigned int idx) +{ + int err; + struct snd_card *card; + struct snd_msnd *chip; + + printk(KERN_INFO LOGNAME ": Turtle Beach " LONGNAME " Linux Driver Version " + VERSION ", Copyright (C) 2002 Karsten Wiese 1998 Andrew Veliath\n"); + + if (isapnp[idx] || cfg[idx] == SNDRV_AUTO_PORT) { + printk(KERN_INFO LOGNAME ": Assuming PnP mode\n"); + return -ENODEV; + } + + card = snd_card_new(index[idx], id[idx], THIS_MODULE, sizeof(struct snd_msnd)); + if (!card) + return -ENOMEM; + + snd_card_set_dev(card, pdev); + chip = card->private_data; + chip->card = card; + +#ifdef MSND_CLASSIC + switch (irq[idx]) { + case 5: + chip->irqid = HPIRQ_5; break; + case 7: + chip->irqid = HPIRQ_7; break; + case 9: + chip->irqid = HPIRQ_9; break; + case 10: + chip->irqid = HPIRQ_10; break; + case 11: + chip->irqid = HPIRQ_11; break; + case 12: + chip->irqid = HPIRQ_12; break; + } + + switch (mem[idx]) { + case 0xb0000: + chip->memid = HPMEM_B000; break; + case 0xc8000: + chip->memid = HPMEM_C800; break; + case 0xd0000: + chip->memid = HPMEM_D000; break; + case 0xd8000: + chip->memid = HPMEM_D800; break; + case 0xe0000: + chip->memid = HPMEM_E000; break; + case 0xe8000: + chip->memid = HPMEM_E800; break; + } +#else + printk(KERN_INFO LOGNAME ": Non-PnP mode: configuring at port 0x%lx\n", cfg[idx]); + + if (!request_region(cfg[idx], 2, "Pinnacle/Fiji Config")) { + printk(KERN_ERR LOGNAME ": Config port 0x%lx conflict\n", cfg[idx]); + snd_card_free(card); + return -EIO; + } + if (reset[idx]) + if (snd_msnd_pinnacle_cfg_reset(cfg[idx])) { + err = -EIO; + goto cfg_error; + } + + /* DSP */ + err = snd_msnd_write_cfg_logical(cfg[idx], 0, + io[idx], 0, + irq[idx], mem[idx]); + + if (err) + goto cfg_error; + + /* The following are Pinnacle specific */ + + /* MPU */ + if (mpu_io[idx] != SNDRV_AUTO_PORT + && mpu_irq[idx] != SNDRV_AUTO_IRQ) { + printk(KERN_INFO LOGNAME + ": Configuring MPU to I/O 0x%lx IRQ %d\n", + mpu_io[idx], mpu_irq[idx]); + err = snd_msnd_write_cfg_logical(cfg[idx], 1, + mpu_io[idx], 0, + mpu_irq[idx], 0); + + if (err) + goto cfg_error; + } + + /* IDE */ + if (ide_io0[idx] != SNDRV_AUTO_PORT + && ide_io1[idx] != SNDRV_AUTO_PORT + && ide_irq[idx] != SNDRV_AUTO_IRQ) { + printk(KERN_INFO LOGNAME + ": Configuring IDE to I/O 0x%lx, 0x%lx IRQ %d\n", + ide_io0[idx], ide_io1[idx], ide_irq[idx]); + err = snd_msnd_write_cfg_logical(cfg[idx], 2, + ide_io0[idx], ide_io1[idx], + ide_irq[idx], 0); + + if (err) + goto cfg_error; + } + + /* Joystick */ + if (joystick_io[idx] != SNDRV_AUTO_PORT) { + printk(KERN_INFO LOGNAME + ": Configuring joystick to I/O 0x%lx\n", + joystick_io[idx]); + err = snd_msnd_write_cfg_logical(cfg[idx], 3, + joystick_io[idx], 0, + 0, 0); + + if (err) + goto cfg_error; + } + release_region(cfg[idx], 2); + +#endif /* MSND_CLASSIC */ + + set_default_audio_parameters(chip); +#ifdef MSND_CLASSIC + chip->type = msndClassic; +#else + chip->type = msndPinnacle; +#endif +/* snd_msnd_playback.buffer_bytes_max = 3 * ( + snd_msnd_playback.period_bytes_max = + snd_msnd_playback.period_bytes_min = + chip->play_period_bytes = play_period_bytes + ); + printk(KERN_INFO LOGNAME ": play_period_bytes=0x%X\n", chip->play_period_bytes); +*/ + chip->io = io[idx]; + chip->irq = irq[idx]; + chip->base = mem[idx]; + + chip->calibrate_signal = calibrate_signal ? 1 : 0; + chip->recsrc = 0; + chip->dspq_data_buff = DSPQ_DATA_BUFF; + chip->dspq_buff_size = DSPQ_BUFF_SIZE; + if (write_ndelay[idx]) + clear_bit(F_DISABLE_WRITE_NDELAY, &chip->flags); + else + set_bit(F_DISABLE_WRITE_NDELAY, &chip->flags); +#ifndef MSND_CLASSIC + if (digital[idx]) + set_bit(F_HAVEDIGITAL, &chip->flags); +#endif + spin_lock_init(&chip->lock); + err = snd_msnd_probe(card); + if (err < 0) { + printk(KERN_ERR LOGNAME ": Probe failed\n"); + snd_card_free(card); + return err; + } + + err = snd_msnd_attach(card); + if (err < 0) { + printk(KERN_ERR LOGNAME ": Attach failed\n"); + snd_card_free(card); + return err; + } + dev_set_drvdata(pdev, card); + + return 0; + +#ifndef MSND_CLASSIC +cfg_error: + release_region(cfg[idx], 2); + snd_card_free(card); + return err; +#endif +} + +static int __devexit snd_msnd_isa_remove(struct device *pdev, unsigned int dev) +{ + snd_msnd_unload(dev_get_drvdata(pdev)); + dev_set_drvdata(pdev, NULL); + return 0; +} + +#define DEV_NAME "msnd-pinnacle" + +static struct isa_driver snd_msnd_driver = { + .match = snd_msnd_isa_match, + .probe = snd_msnd_isa_probe, + .remove = __devexit_p(snd_msnd_isa_remove), + /* FIXME: suspend, resume */ + .driver = { + .name = DEV_NAME + }, +}; + +#ifdef CONFIG_PNP +static int __devinit snd_msnd_pnp_detect(struct pnp_card_link *pcard, + const struct pnp_card_device_id *pid) +{ + static int idx = 0; + struct pnp_dev *pnp_dev; + struct pnp_dev *mpu_dev; + struct snd_card *card; + struct snd_msnd *chip; + int ret; + + for ( ; idx < SNDRV_CARDS; idx++) { + if (isapnp[idx]) + break; + } + if (idx >= SNDRV_CARDS) + return -ENODEV; + + /* + * Check that we still have room for another sound card ... + */ + pnp_dev = pnp_request_card_device(pcard, pid->devs[0].id, NULL); + if (!pnp_dev) + return -ENODEV; + + mpu_dev = pnp_request_card_device(pcard, pid->devs[1].id, NULL); + if (!mpu_dev) + return -ENODEV; + + if (!pnp_is_active(pnp_dev) && pnp_activate_dev(pnp_dev) < 0) { + printk(KERN_INFO "msnd_pinnacle: device is inactive\n"); + return -EBUSY; + } + + if (!pnp_is_active(mpu_dev) && pnp_activate_dev(mpu_dev) < 0) { + printk(KERN_INFO "msnd_pinnacle: MPU device is inactive\n"); + return -EBUSY; + } + + /* + * Create a new ALSA sound card entry, in anticipation + * of detecting our hardware ... + */ + card = snd_card_new(index[idx], id[idx], THIS_MODULE, + sizeof(struct snd_msnd)); + if (!card) + return -ENOMEM; + + chip = card->private_data; + chip->card = card; + snd_card_set_dev(card, &pcard->card->dev); + + /* + * Read the correct parameters off the ISA PnP bus ... + */ + io[idx] = pnp_port_start(pnp_dev, 0); + irq[idx] = pnp_irq(pnp_dev, 0); + mem[idx] = pnp_mem_start(pnp_dev, 0); + mpu_io[idx] = pnp_port_start(mpu_dev, 0); + mpu_irq[idx] = pnp_irq(mpu_dev, 0); + + set_default_audio_parameters(chip); +#ifdef MSND_CLASSIC + chip->type = msndClassic; +#else + chip->type = msndPinnacle; +#endif + chip->io = io[idx]; + chip->irq = irq[idx]; + chip->base = mem[idx]; + + chip->calibrate_signal = calibrate_signal ? 1 : 0; + chip->recsrc = 0; + chip->dspq_data_buff = DSPQ_DATA_BUFF; + chip->dspq_buff_size = DSPQ_BUFF_SIZE; + if (write_ndelay[idx]) + clear_bit(F_DISABLE_WRITE_NDELAY, &chip->flags); + else + set_bit(F_DISABLE_WRITE_NDELAY, &chip->flags); +#ifndef MSND_CLASSIC + if (digital[idx]) + set_bit(F_HAVEDIGITAL, &chip->flags); +#endif + spin_lock_init(&chip->lock); + ret = snd_msnd_probe(card); + if (ret < 0) { + printk(KERN_ERR LOGNAME ": Probe failed\n"); + goto _release_card; + } + + ret = snd_msnd_attach(card); + if (ret < 0) { + printk(KERN_ERR LOGNAME ": Attach failed\n"); + goto _release_card; + } + + pnp_set_card_drvdata(pcard, card); + ++idx; + return 0; + +_release_card: + snd_card_free(card); + return ret; +} + +static void __devexit snd_msnd_pnp_remove(struct pnp_card_link *pcard) +{ + snd_msnd_unload(pnp_get_card_drvdata(pcard)); + pnp_set_card_drvdata(pcard, NULL); +} + +static int isa_registered; +static int pnp_registered; + +static struct pnp_card_device_id msnd_pnpids[] = { + /* Pinnacle PnP */ + { .id = "BVJ0440", .devs = { { "TBS0000" }, { "TBS0001" } } }, + { .id = "" } /* end */ +}; + +MODULE_DEVICE_TABLE(pnp_card, msnd_pnpids); + +static struct pnp_card_driver msnd_pnpc_driver = { + .flags = PNP_DRIVER_RES_DO_NOT_CHANGE, + .name = "msnd_pinnacle", + .id_table = msnd_pnpids, + .probe = snd_msnd_pnp_detect, + .remove = __devexit_p(snd_msnd_pnp_remove), +}; +#endif /* CONFIG_PNP */ + +static int __init snd_msnd_init(void) +{ + int err; + + err = isa_register_driver(&snd_msnd_driver, SNDRV_CARDS); +#ifdef CONFIG_PNP + if (!err) + isa_registered = 1; + + err = pnp_register_card_driver(&msnd_pnpc_driver); + if (!err) + pnp_registered = 1; + + if (isa_registered) + err = 0; +#endif + return err; +} + +static void __exit snd_msnd_exit(void) +{ +#ifdef CONFIG_PNP + if (pnp_registered) + pnp_unregister_card_driver(&msnd_pnpc_driver); + if (isa_registered) +#endif + isa_unregister_driver(&snd_msnd_driver); +} + +module_init(snd_msnd_init); +module_exit(snd_msnd_exit); + diff --git a/sound/isa/msnd/msnd_pinnacle.h b/sound/isa/msnd/msnd_pinnacle.h new file mode 100644 index 0000000..b6c400e --- /dev/null +++ b/sound/isa/msnd/msnd_pinnacle.h @@ -0,0 +1,246 @@ +/********************************************************************* + * + * msnd_pinnacle.h + * + * Turtle Beach MultiSound Sound Card Driver for Linux + * + * Some parts of this header file were derived from the Turtle Beach + * MultiSound Driver Development Kit. + * + * Copyright (C) 1998 Andrew Veliath + * Copyright (C) 1993 Turtle Beach Systems, Inc. + * + * 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., 675 Mass Ave, Cambridge, MA 02139, USA. + * + ********************************************************************/ +#ifndef __MSND_PINNACLE_H +#define __MSND_PINNACLE_H + +#define DSP_NUMIO 0x08 + +#define IREG_LOGDEVICE 0x07 +#define IREG_ACTIVATE 0x30 +#define LD_ACTIVATE 0x01 +#define LD_DISACTIVATE 0x00 +#define IREG_EECONTROL 0x3F +#define IREG_MEMBASEHI 0x40 +#define IREG_MEMBASELO 0x41 +#define IREG_MEMCONTROL 0x42 +#define IREG_MEMRANGEHI 0x43 +#define IREG_MEMRANGELO 0x44 +#define MEMTYPE_8BIT 0x00 +#define MEMTYPE_16BIT 0x02 +#define MEMTYPE_RANGE 0x00 +#define MEMTYPE_HIADDR 0x01 +#define IREG_IO0_BASEHI 0x60 +#define IREG_IO0_BASELO 0x61 +#define IREG_IO1_BASEHI 0x62 +#define IREG_IO1_BASELO 0x63 +#define IREG_IRQ_NUMBER 0x70 +#define IREG_IRQ_TYPE 0x71 +#define IRQTYPE_HIGH 0x02 +#define IRQTYPE_LOW 0x00 +#define IRQTYPE_LEVEL 0x01 +#define IRQTYPE_EDGE 0x00 + +#define HP_DSPR 0x04 +#define HP_BLKS 0x04 + +#define HPDSPRESET_OFF 2 +#define HPDSPRESET_ON 0 + +#define HPBLKSEL_0 2 +#define HPBLKSEL_1 3 + +#define HIMT_DAT_OFF 0x03 + +#define HIDSP_PLAY_UNDER 0x00 +#define HIDSP_INT_PLAY_UNDER 0x01 +#define HIDSP_SSI_TX_UNDER 0x02 +#define HIDSP_RECQ_OVERFLOW 0x08 +#define HIDSP_INT_RECORD_OVER 0x09 +#define HIDSP_SSI_RX_OVERFLOW 0x0a + +#define HIDSP_MIDI_IN_OVER 0x10 + +#define HIDSP_MIDI_FRAME_ERR 0x11 +#define HIDSP_MIDI_PARITY_ERR 0x12 +#define HIDSP_MIDI_OVERRUN_ERR 0x13 + +#define HIDSP_INPUT_CLIPPING 0x20 +#define HIDSP_MIX_CLIPPING 0x30 +#define HIDSP_DAT_IN_OFF 0x21 + +#define HDEXAR_SET_ANA_IN 0 +#define HDEXAR_CLEAR_PEAKS 1 +#define HDEXAR_IN_SET_POTS 2 +#define HDEXAR_AUX_SET_POTS 3 +#define HDEXAR_CAL_A_TO_D 4 +#define HDEXAR_RD_EXT_DSP_BITS 5 + +#define HDEXAR_SET_SYNTH_IN 4 +#define HDEXAR_READ_DAT_IN 5 +#define HDEXAR_MIC_SET_POTS 6 +#define HDEXAR_SET_DAT_IN 7 + +#define HDEXAR_SET_SYNTH_48 8 +#define HDEXAR_SET_SYNTH_44 9 + +#define TIME_PRO_RESET_DONE 0x028A +#define TIME_PRO_SYSEX 0x001E +#define TIME_PRO_RESET 0x0032 + +#define AGND 0x01 +#define SIGNAL 0x02 + +#define EXT_DSP_BIT_DCAL 0x0001 +#define EXT_DSP_BIT_MIDI_CON 0x0002 + +#define BUFFSIZE 0x8000 +#define HOSTQ_SIZE 0x40 + +#define SRAM_CNTL_START 0x7F00 +#define SMA_STRUCT_START 0x7F40 + +#define DAP_BUFF_SIZE 0x2400 +#define DAR_BUFF_SIZE 0x1000 + +#define DAPQ_STRUCT_SIZE 0x10 +#define DARQ_STRUCT_SIZE 0x10 +#define DAPQ_BUFF_SIZE (3 * 0x10) +#define DARQ_BUFF_SIZE (3 * 0x10) +#define MODQ_BUFF_SIZE 0x400 +#define MIDQ_BUFF_SIZE 0x800 +#define DSPQ_BUFF_SIZE 0x5A0 + +#define DAPQ_DATA_BUFF 0x6C00 +#define DARQ_DATA_BUFF 0x6C30 +#define MODQ_DATA_BUFF 0x6C60 +#define MIDQ_DATA_BUFF 0x7060 +#define DSPQ_DATA_BUFF 0x7860 + +#define DAPQ_OFFSET SRAM_CNTL_START +#define DARQ_OFFSET (SRAM_CNTL_START + 0x08) +#define MODQ_OFFSET (SRAM_CNTL_START + 0x10) +#define MIDQ_OFFSET (SRAM_CNTL_START + 0x18) +#define DSPQ_OFFSET (SRAM_CNTL_START + 0x20) + +#define MOP_WAVEHDR 0 +#define MOP_EXTOUT 1 +#define MOP_HWINIT 0xfe +#define MOP_NONE 0xff +#define MOP_MAX 1 + +#define MIP_EXTIN 0 +#define MIP_WAVEHDR 1 +#define MIP_HWINIT 0xfe +#define MIP_MAX 1 + +/* Pinnacle/Fiji SMA Common Data */ +#define SMA_wCurrPlayBytes 0x0000 +#define SMA_wCurrRecordBytes 0x0002 +#define SMA_wCurrPlayVolLeft 0x0004 +#define SMA_wCurrPlayVolRight 0x0006 +#define SMA_wCurrInVolLeft 0x0008 +#define SMA_wCurrInVolRight 0x000a +#define SMA_wCurrMHdrVolLeft 0x000c +#define SMA_wCurrMHdrVolRight 0x000e +#define SMA_dwCurrPlayPitch 0x0010 +#define SMA_dwCurrPlayRate 0x0014 +#define SMA_wCurrMIDIIOPatch 0x0018 +#define SMA_wCurrPlayFormat 0x001a +#define SMA_wCurrPlaySampleSize 0x001c +#define SMA_wCurrPlayChannels 0x001e +#define SMA_wCurrPlaySampleRate 0x0020 +#define SMA_wCurrRecordFormat 0x0022 +#define SMA_wCurrRecordSampleSize 0x0024 +#define SMA_wCurrRecordChannels 0x0026 +#define SMA_wCurrRecordSampleRate 0x0028 +#define SMA_wCurrDSPStatusFlags 0x002a +#define SMA_wCurrHostStatusFlags 0x002c +#define SMA_wCurrInputTagBits 0x002e +#define SMA_wCurrLeftPeak 0x0030 +#define SMA_wCurrRightPeak 0x0032 +#define SMA_bMicPotPosLeft 0x0034 +#define SMA_bMicPotPosRight 0x0035 +#define SMA_bMicPotMaxLeft 0x0036 +#define SMA_bMicPotMaxRight 0x0037 +#define SMA_bInPotPosLeft 0x0038 +#define SMA_bInPotPosRight 0x0039 +#define SMA_bAuxPotPosLeft 0x003a +#define SMA_bAuxPotPosRight 0x003b +#define SMA_bInPotMaxLeft 0x003c +#define SMA_bInPotMaxRight 0x003d +#define SMA_bAuxPotMaxLeft 0x003e +#define SMA_bAuxPotMaxRight 0x003f +#define SMA_bInPotMaxMethod 0x0040 +#define SMA_bAuxPotMaxMethod 0x0041 +#define SMA_wCurrMastVolLeft 0x0042 +#define SMA_wCurrMastVolRight 0x0044 +#define SMA_wCalFreqAtoD 0x0046 +#define SMA_wCurrAuxVolLeft 0x0048 +#define SMA_wCurrAuxVolRight 0x004a +#define SMA_wCurrPlay1VolLeft 0x004c +#define SMA_wCurrPlay1VolRight 0x004e +#define SMA_wCurrPlay2VolLeft 0x0050 +#define SMA_wCurrPlay2VolRight 0x0052 +#define SMA_wCurrPlay3VolLeft 0x0054 +#define SMA_wCurrPlay3VolRight 0x0056 +#define SMA_wCurrPlay4VolLeft 0x0058 +#define SMA_wCurrPlay4VolRight 0x005a +#define SMA_wCurrPlay1PeakLeft 0x005c +#define SMA_wCurrPlay1PeakRight 0x005e +#define SMA_wCurrPlay2PeakLeft 0x0060 +#define SMA_wCurrPlay2PeakRight 0x0062 +#define SMA_wCurrPlay3PeakLeft 0x0064 +#define SMA_wCurrPlay3PeakRight 0x0066 +#define SMA_wCurrPlay4PeakLeft 0x0068 +#define SMA_wCurrPlay4PeakRight 0x006a +#define SMA_wCurrPlayPeakLeft 0x006c +#define SMA_wCurrPlayPeakRight 0x006e +#define SMA_wCurrDATSR 0x0070 +#define SMA_wCurrDATRXCHNL 0x0072 +#define SMA_wCurrDATTXCHNL 0x0074 +#define SMA_wCurrDATRXRate 0x0076 +#define SMA_dwDSPPlayCount 0x0078 +#define SMA__size 0x007c + +#ifdef HAVE_DSPCODEH +# include "pndsperm.c" +# include "pndspini.c" +# define PERMCODE pndsperm +# define INITCODE pndspini +# define PERMCODESIZE sizeof(pndsperm) +# define INITCODESIZE sizeof(pndspini) +#else +# ifndef CONFIG_MSNDPIN_INIT_FILE +# define CONFIG_MSNDPIN_INIT_FILE "turtlebeach/pndspini.bin" +# endif +# ifndef CONFIG_MSNDPIN_PERM_FILE +# define CONFIG_MSNDPIN_PERM_FILE "turtlebeach/pndsperm.bin" +# endif +# define PERMCODEFILE CONFIG_MSNDPIN_PERM_FILE +# define INITCODEFILE CONFIG_MSNDPIN_INIT_FILE +# define PERMCODE dspini +# define INITCODE permini +# define PERMCODESIZE sizeof_dspini +# define INITCODESIZE sizeof_permini +#endif +#define LONGNAME "MultiSound (Pinnacle/Fiji)" + +int snd_msnd_send_dsp_cmd_chk(struct snd_msnd *chip, register u8 cmd); + + +#endif /* __MSND_PINNACLE_H */ diff --git a/sound/isa/msnd/msnd_pinnacle_mixer.c b/sound/isa/msnd/msnd_pinnacle_mixer.c new file mode 100644 index 0000000..de94e75 --- /dev/null +++ b/sound/isa/msnd/msnd_pinnacle_mixer.c @@ -0,0 +1,343 @@ +/*************************************************************************** + msnd_pinnacle_mixer.c - description + ------------------- + begin : Fre Jun 7 2002 + copyright : (C) 2002 by karsten wiese + email : annabellesgarden(a)yahoo.de + ***************************************************************************/ + +/*************************************************************************** + * * + * This program is free software; you can redistribute it and/or modify * + * it under the terms of the GNU General Public License as published by * + * the Free Software Foundation; either version 2 of the License, or * + * (at your option) any later version. * + * * + ***************************************************************************/ + +#include <linux/init.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/control.h> +#include "msnd.h" +#include "msnd_pinnacle.h" + + +#define MSND_MIXER_VOLUME 0 +#define MSND_MIXER_PCM 1 +#define MSND_MIXER_AUX 2 /* Input source 1 (aux1) */ +#define MSND_MIXER_IMIX 3 /* Recording monitor */ +#define MSND_MIXER_SYNTH 4 +#define MSND_MIXER_SPEAKER 5 +#define MSND_MIXER_LINE 6 +#define MSND_MIXER_MIC 7 +#define MSND_MIXER_RECLEV 11 /* Recording level */ +#define MSND_MIXER_IGAIN 12 /* Input gain */ +#define MSND_MIXER_OGAIN 13 /* Output gain */ +#define MSND_MIXER_DIGITAL 17 /* Digital (input) 1 */ + +/* Device mask bits */ + +#define MSND_MASK_VOLUME (1 << MSND_MIXER_VOLUME) +#define MSND_MASK_SYNTH (1 << MSND_MIXER_SYNTH) +#define MSND_MASK_PCM (1 << MSND_MIXER_PCM) +#define MSND_MASK_SPEAKER (1 << MSND_MIXER_SPEAKER) +#define MSND_MASK_LINE (1 << MSND_MIXER_LINE) +#define MSND_MASK_MIC (1 << MSND_MIXER_MIC) +#define MSND_MASK_IMIX (1 << MSND_MIXER_IMIX) +#define MSND_MASK_RECLEV (1 << MSND_MIXER_RECLEV) +#define MSND_MASK_IGAIN (1 << MSND_MIXER_IGAIN) +#define MSND_MASK_OGAIN (1 << MSND_MIXER_OGAIN) +#define MSND_MASK_AUX (1 << MSND_MIXER_AUX) +#define MSND_MASK_DIGITAL (1 << MSND_MIXER_DIGITAL) + + + + + +static int snd_msndmix_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + static char *texts[3] = { + "Analog", "SPDIF", "MASS" + }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 3; + if (uinfo->value.enumerated.item > 2) + uinfo->value.enumerated.item = 2; + strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); + return 0; +} + +static int snd_msndmix_get_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_msnd *msnd = snd_kcontrol_chip(kcontrol); + ucontrol->value.enumerated.item[0] = 0; + +/* if (msnd->recsrc & MSND_MASK_IMIX) { this is the default + ucontrol->value.enumerated.item[0] = 0; + } + else */if (msnd->recsrc & MSND_MASK_SYNTH) { + ucontrol->value.enumerated.item[0] = 2; + } + else if ((msnd->recsrc & MSND_MASK_DIGITAL) && test_bit(F_HAVEDIGITAL, &msnd->flags)) { + ucontrol->value.enumerated.item[0] = 1; + } + + return 0; +} + + +static int snd_msndmix_set_mux(struct snd_msnd *msnd, int val) +{ + unsigned newrecsrc; + int change; + unsigned char msndbyte; + + switch (val) { + case 0: + newrecsrc = MSND_MASK_IMIX; + msndbyte = HDEXAR_SET_ANA_IN; + break; + case 1: + newrecsrc = MSND_MASK_DIGITAL; + msndbyte = HDEXAR_SET_DAT_IN; + break; + case 2: + newrecsrc = MSND_MASK_SYNTH; + msndbyte = HDEXAR_SET_SYNTH_IN; + break; + default: + return -EINVAL; + } + change = newrecsrc != msnd->recsrc; + if (change) { + if (0 == snd_msnd_send_word(msnd, 0, 0, msndbyte)) + if (0 == snd_msnd_send_dsp_cmd_chk(msnd, HDEX_AUX_REQ)) + msnd->recsrc = newrecsrc; + } + return change; +} + + +static int snd_msndmix_put_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_msnd *msnd = snd_kcontrol_chip(kcontrol); + return snd_msndmix_set_mux(msnd, ucontrol->value.enumerated.item[0]); +} + + +static int snd_msndmix_volume_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 100; + return 0; +} + +static int snd_msndmix_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_msnd *msnd = snd_kcontrol_chip(kcontrol); + int addr = kcontrol->private_value; + + ucontrol->value.integer.value[0] = (msnd->left_levels[addr] * 100) / 0xFFFF; + ucontrol->value.integer.value[1] = (msnd->right_levels[addr] * 100) / 0xFFFF; + return 0; +} + + +#define update_volm(a,b) \ + isa_writew((dev->left_levels[a] >> 1) * \ + isa_readw(dev->SMA + SMA_wCurrMastVolLeft) / 0xffff, \ + dev->SMA + SMA_##b##Left); \ + isa_writew((dev->right_levels[a] >> 1) * \ + isa_readw(dev->SMA + SMA_wCurrMastVolRight) / 0xffff, \ + dev->SMA + SMA_##b##Right); + +#define update_potm(d,s,ar) \ + isa_writeb((dev->left_levels[d] >> 8) * \ + isa_readw(dev->SMA + SMA_wCurrMastVolLeft) / 0xffff, \ + dev->SMA + SMA_##s##Left); \ + isa_writeb((dev->right_levels[d] >> 8) * \ + isa_readw(dev->SMA + SMA_wCurrMastVolRight) / 0xffff, \ + dev->SMA + SMA_##s##Right); \ + if (snd_msnd_send_word(dev, 0, 0, ar) == 0) \ + snd_msnd_send_dsp_cmd_chk(dev, HDEX_AUX_REQ); + +#define update_pot(d,s,ar) \ + isa_writeb(dev->left_levels[d] >> 8, \ + dev->SMA + SMA_##s##Left); \ + isa_writeb(dev->right_levels[d] >> 8, \ + dev->SMA + SMA_##s##Right); \ + if (snd_msnd_send_word(dev, 0, 0, ar) == 0) \ + snd_msnd_send_dsp_cmd_chk(dev, HDEX_AUX_REQ); + + +static int snd_msndmix_set(struct snd_msnd *dev, int d, int left, int right) +{ + int bLeft, bRight; + int wLeft, wRight; + int updatemaster = 0; + + //snd_printd("mixer_set(struct snd_msnd * %X, d=%i, left=%i, right=%i\n", + // (unsigned)dev, d, left, right); + + if (d >= LEVEL_ENTRIES) + return -EINVAL; + + bLeft = left * 0xff / 100; + wLeft = left * 0xffff / 100; + + bRight = right * 0xff / 100; + wRight = right * 0xffff / 100; + + dev->left_levels[d] = wLeft; + dev->right_levels[d] = wRight; + + switch (d) { + /* master volume unscaled controls */ + case MSND_MIXER_LINE: /* line pot control */ + /* scaled by IMIX in digital mix */ + isa_writeb(bLeft, dev->SMA + SMA_bInPotPosLeft); + isa_writeb(bRight, dev->SMA + SMA_bInPotPosRight); + if (snd_msnd_send_word(dev, 0, 0, HDEXAR_IN_SET_POTS) == 0) + snd_msnd_send_dsp_cmd_chk(dev, HDEX_AUX_REQ); + break; +#ifndef MSND_CLASSIC + case MSND_MIXER_MIC: /* mic pot control */ + /* scaled by IMIX in digital mix */ + isa_writeb(bLeft, dev->SMA + SMA_bMicPotPosLeft); + isa_writeb(bRight, dev->SMA + SMA_bMicPotPosRight); + if (snd_msnd_send_word(dev, 0, 0, HDEXAR_MIC_SET_POTS) == 0) + snd_msnd_send_dsp_cmd_chk(dev, HDEX_AUX_REQ); + break; +#endif + case MSND_MIXER_VOLUME: /* master volume */ + isa_writew(wLeft, dev->SMA + SMA_wCurrMastVolLeft); + isa_writew(wRight, dev->SMA + SMA_wCurrMastVolRight); + /* fall through */ + + case MSND_MIXER_AUX: /* aux pot control */ + /* scaled by master volume */ + /* fall through */ + + /* digital controls */ + case MSND_MIXER_SYNTH: /* synth vol (dsp mix) */ + case MSND_MIXER_PCM: /* pcm vol (dsp mix) */ + case MSND_MIXER_IMIX: /* input monitor (dsp mix) */ + /* scaled by master volume */ + updatemaster = 1; + break; + + default: + return 0; + } + + if (updatemaster) { + /* update master volume scaled controls */ + update_volm(MSND_MIXER_PCM, wCurrPlayVol); + update_volm(MSND_MIXER_IMIX, wCurrInVol); +#ifndef MSND_CLASSIC + update_volm(MSND_MIXER_SYNTH, wCurrMHdrVol); +#endif + update_potm(MSND_MIXER_AUX, bAuxPotPos, HDEXAR_AUX_SET_POTS); + } + + return 0; +} + + +static int snd_msndmix_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_msnd *msnd = snd_kcontrol_chip(kcontrol); + int change, addr = kcontrol->private_value; + int left, right; + // unsigned long flags; + + left = ucontrol->value.integer.value[0] % 101; + right = ucontrol->value.integer.value[1] % 101; + // spin_lock_irqsave(&msnd->mixer_lock, flags); + change = msnd->left_levels[addr] != left + || msnd->right_levels[addr] != right; + snd_msndmix_set(msnd, addr, left, right); + // spin_unlock__irqrestore(&msnd->mixer_lock, flags); + return change; +} + + +#define DUMMY_VOLUME(xname, xindex, addr) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .info = snd_msndmix_volume_info, \ + .get = snd_msndmix_volume_get, .put = snd_msndmix_volume_put, \ + .private_value = addr } + + +static struct snd_kcontrol_new snd_msnd_controls[] = { +DUMMY_VOLUME("Master Volume", 0, MSND_MIXER_VOLUME), +DUMMY_VOLUME("PCM Volume", 0, MSND_MIXER_PCM), +DUMMY_VOLUME("Aux Volume", 0, MSND_MIXER_AUX), +DUMMY_VOLUME("Line Volume", 0, MSND_MIXER_LINE), +DUMMY_VOLUME("Mic Volume", 0, MSND_MIXER_MIC), +DUMMY_VOLUME("Monitor", 0, MSND_MIXER_IMIX), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = snd_msndmix_info_mux, + .get = snd_msndmix_get_mux, + .put = snd_msndmix_put_mux, +} +}; + + + +int __devinit snd_msndmix_new(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + unsigned int idx; + int err; + + if (snd_BUG_ON(!chip)) + return -EINVAL; + // spin_lock_init(&chip->mixer_lock); + strcpy(card->mixername, "MSND Pinnacle Mixer"); + + for (idx = 0; idx < ARRAY_SIZE(snd_msnd_controls); idx++) + err = snd_ctl_add(card, snd_ctl_new1(snd_msnd_controls + idx, chip)); + if (err < 0) + return err; + +#ifndef MSND_CLASSIC + // snd_msndmix_force_recsrc(chip, 0); +#endif + + return 0; +} + + + +void snd_msndmix_setup(struct snd_msnd *dev) +{ + update_pot(MSND_MIXER_LINE, bInPotPos, HDEXAR_IN_SET_POTS); + update_potm(MSND_MIXER_AUX, bAuxPotPos, HDEXAR_AUX_SET_POTS); + update_volm(MSND_MIXER_PCM, wCurrPlayVol); + update_volm(MSND_MIXER_IMIX, wCurrInVol); +#ifndef MSND_CLASSIC + update_pot(MSND_MIXER_MIC, bMicPotPos, HDEXAR_MIC_SET_POTS); + update_volm(MSND_MIXER_SYNTH, wCurrMHdrVol); +#endif + +} + + +unsigned long snd_msndmix_force_recsrc(struct snd_msnd *dev, int recsrc) +{ +// snd_msndmix_set(dev, MSND_MIXER_VOLUME, 100, 100); +// snd_msndmix_set(dev, MSND_MIXER_LINE, 70, 70); +// snd_msndmix_set(dev, MSND_MIXER_IMIX, 100, 100); + dev->recsrc = -1; + return snd_msndmix_set_mux(dev, recsrc); +} + + ---------------------------------------------------------------------- Szukasz mieszkania, domu, dzialki? Sprawdz >>> http://link.interia.pl/f1f96

2 4

[alsa-devel] [PATCH] hda: Add PCI DFI vendor ID
by Matthew Ranostay 22 Jan '09

22 Jan '09

Add a define for DFI PCI vendor id. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/include/linux/pci_ids.h b/include/linux/pci_ids.h index d56ad9c..c33d50c 100644 --- a/include/linux/pci_ids.h +++ b/include/linux/pci_ids.h @@ -2106,6 +2106,8 @@ #define PCI_DEVICE_ID_MELLANOX_SINAI_OLD 0x5e8c #define PCI_DEVICE_ID_MELLANOX_SINAI 0x6274 +#define PCI_VENDOR_ID_DFI 0x15bd + #define PCI_VENDOR_ID_QUICKNET 0x15e2 #define PCI_DEVICE_ID_QUICKNET_XJ 0x0500

1 0

Re: [alsa-devel] [PATCH 1/2] Remove definition for surround sound devices on ICE1712.
by John Rigg 22 Jan '09

22 Jan '09

On Tue, Jan 20, 2009 at 02:49:53PM +0100, Diego E. 'Flameeyes' Pettenò wrote: > Jaroslav Kysela <perex(a)perex.cz> writes: > > > Do you have seen M-Audio Delta 1010 ? The problem is that we have hardware > > that can be used for surround output, but alsa-lib configuration files > > cannot describe the restrictions in detail now. > > I thought that was an ICE1724, rather than 1712; I was mistaken, sorry. > > Still, as they are the devices are non-functional at least on my card, > and I'm ready to bet that the same would happen on the Delta 1010 given > that the number of channels is not coherent (you try to open surround71 > as 8-channels and it'll fail because this minimal card has 10; if the > 1010 has the same amount of channels, it'll still fail). The ice1712 has 12 inputs and 10 outputs, regardless of how many physical in/outs your card has. Most of the ice1712 interfaces I've seen have 8 analogue in/outs along with S/PDIF in/out. I can't think of many stereo-only ice1712 cards (apart from the Audiophile 2496). John

1 0

[alsa-devel] [PATCH] do not create OPL3 timers if there is no OPL3 irq wired
by Krzysztof Helt 21 Jan '09

21 Jan '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Most cards have OPL3 FM synthetiser but they do not have OPL3 interrupt wired to a sound chip or CPU. Do not create OPL3 timers for such cards as the timers are useless witthout interrupt. This patch removes OPL3 timers for following alsa drivers: snd-ad1816a, snd-opti93x, snd-opti92x, snd-sc6000, snd-cmi8330. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/ad1816a/ad1816a.c | 7 ++----- sound/isa/cmi8330.c | 4 ---- sound/isa/opti9xx/opti92x-ad1848.c | 10 ++-------- sound/isa/sc6000.c | 4 ---- 4 files changed, 4 insertions(+), 21 deletions(-) diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c index 7752424..3810833 100644 --- a/sound/isa/ad1816a/ad1816a.c +++ b/sound/isa/ad1816a/ad1816a.c @@ -207,11 +207,8 @@ static int __devinit snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard OPL3_HW_AUTO, 0, &opl3) < 0) { printk(KERN_ERR PFX "no OPL device at 0x%lx-0x%lx.\n", fm_port[dev], fm_port[dev] + 2); } else { - if ((error = snd_opl3_timer_new(opl3, 1, 2)) < 0) { - snd_card_free(card); - return error; - } - if ((error = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) { + error = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (error < 0) { snd_card_free(card); return error; } diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index dec6ea5..1154379 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -546,10 +546,6 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev) "no OPL device at 0x%lx-0x%lx ?\n", fmport[dev], fmport[dev] + 2); } else { - err = snd_opl3_timer_new(opl3, 0, 1); - if (err < 0) - return err; - err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); if (err < 0) return err; diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 19706b0..5deb7e6 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -815,14 +815,8 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) chip->fm_port, chip->fm_port + 4 - 1); } if (opl3) { -#ifdef CS4231 - const int t1dev = 1; -#else - const int t1dev = 0; -#endif - if ((error = snd_opl3_timer_new(opl3, t1dev, t1dev+1)) < 0) - return error; - if ((error = snd_opl3_hwdep_new(opl3, 0, 1, &synth)) < 0) + error = snd_opl3_hwdep_new(opl3, 0, 1, &synth); + if (error < 0) return error; } } diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index ca35924..bbc5369 100644 --- a/sound/isa/sc6000.c +++ b/sound/isa/sc6000.c @@ -576,10 +576,6 @@ static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev) snd_printk(KERN_ERR PFX "no OPL device at 0x%x-0x%x ?\n", 0x388, 0x388 + 2); } else { - err = snd_opl3_timer_new(opl3, 0, 1); - if (err < 0) - goto err_unmap2; - err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); if (err < 0) goto err_unmap2; -- 1.5.2.2 ----------------------------------------------------------------------- Promocja w Speak Up. Kwartal angielskiego za darmo. 3 miesiace nauki gratis. Sprawdz teraz! >> http://link.interia.pl/f2019

2 1

[alsa-devel] [PATCH] ad1816a: enable hardware timer
by Krzysztof Helt 21 Jan '09

21 Jan '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Enable hardware timer with 10 usec resolution. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This timer passes the test with the "timer" command from the alsa-lib/test package. Amazing, that a whole code was already in the driver just commented out. include/sound/ad1816a.h | 2 ++ sound/isa/ad1816a/ad1816a.c | 7 +++++++ sound/isa/ad1816a/ad1816a_lib.c | 5 ----- 3 files changed, 9 insertions(+), 5 deletions(-) diff --git a/include/sound/ad1816a.h b/include/sound/ad1816a.h index b3aa62e..d010858 100644 --- a/include/sound/ad1816a.h +++ b/include/sound/ad1816a.h @@ -169,5 +169,7 @@ extern int snd_ad1816a_create(struct snd_card *card, unsigned long port, extern int snd_ad1816a_pcm(struct snd_ad1816a *chip, int device, struct snd_pcm **rpcm); extern int snd_ad1816a_mixer(struct snd_ad1816a *chip); +extern int snd_ad1816a_timer(struct snd_ad1816a *chip, int device, + struct snd_timer **rtimer); #endif /* __SOUND_AD1816A_H */ diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c index 3810833..15f6010 100644 --- a/sound/isa/ad1816a/ad1816a.c +++ b/sound/isa/ad1816a/ad1816a.c @@ -156,6 +156,7 @@ static int __devinit snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard struct snd_card_ad1816a *acard; struct snd_ad1816a *chip; struct snd_opl3 *opl3; + struct snd_timer *timer; if ((card = snd_card_new(index[dev], id[dev], THIS_MODULE, sizeof(struct snd_card_ad1816a))) == NULL) @@ -194,6 +195,12 @@ static int __devinit snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard return error; } + error = snd_ad1816a_timer(chip, 0, &timer); + if (error < 0) { + snd_card_free(card); + return error; + } + if (mpu_port[dev] > 0) { if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, mpu_port[dev], 0, mpu_irq[dev], IRQF_DISABLED, diff --git a/sound/isa/ad1816a/ad1816a_lib.c b/sound/isa/ad1816a/ad1816a_lib.c index 3bfca7c..1c9e01e 100644 --- a/sound/isa/ad1816a/ad1816a_lib.c +++ b/sound/isa/ad1816a/ad1816a_lib.c @@ -377,7 +377,6 @@ static struct snd_pcm_hardware snd_ad1816a_capture = { .fifo_size = 0, }; -#if 0 /* not used now */ static int snd_ad1816a_timer_close(struct snd_timer *timer) { struct snd_ad1816a *chip = snd_timer_chip(timer); @@ -442,8 +441,6 @@ static struct snd_timer_hardware snd_ad1816a_timer_table = { .start = snd_ad1816a_timer_start, .stop = snd_ad1816a_timer_stop, }; -#endif /* not used now */ - static int snd_ad1816a_playback_open(struct snd_pcm_substream *substream) { @@ -687,7 +684,6 @@ int __devinit snd_ad1816a_pcm(struct snd_ad1816a *chip, int device, struct snd_p return 0; } -#if 0 /* not used now */ int __devinit snd_ad1816a_timer(struct snd_ad1816a *chip, int device, struct snd_timer **rtimer) { struct snd_timer *timer; @@ -709,7 +705,6 @@ int __devinit snd_ad1816a_timer(struct snd_ad1816a *chip, int device, struct snd *rtimer = timer; return 0; } -#endif /* not used now */ /* * -- 1.5.2.2 ---------------------------------------------------------------------- Zobacz program TV na dzis! sprawdz >>>http://link.interia.pl/f202a

1 0

[alsa-devel] ASoC: pxa2xx-i2s
by Brian Rhodes 21 Jan '09

21 Jan '09

i2s replay is not being enabled properly on occasion on my pxa270 board. Current code is touching SACR1 in hw_params which I believe is not correct. That should be configured in trigger before enabling i2s. Record is not disabled for playback mode. Any invalid data in the transmit fifo causes deadlock and replay is being disabled. I have been able to reproduce it in kernel 2.6.26 as well as Linus' git tree using speaker-test. (repeatedly starting and stopping). I've verified that this fix prevents (at least) this condition. Please review this patch and let me know if there is anything I have overlooked. Thanks diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 517991f..6cda7f1 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -211,12 +211,10 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, if (!(SACR0 & SACR0_ENB)) { SACR0 = 0; - SACR1 = 0; if (pxa_i2s.master) SACR0 |= SACR0_BCKD; SACR0 |= SACR0_RFTH(14) | SACR0_TFTH(1); - SACR1 |= pxa_i2s.fmt; } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) SAIMR |= SAIMR_TFS; @@ -257,6 +255,12 @@ static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_START: + SACR1 = pxa_i2s.fmt | SACR1_DRPL | SACR1_DREC; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + SACR1 &= ~SACR1_DRPL; + } else { + SACR1 &= ~SACR1_DREC; + } SACR0 |= SACR0_ENB; break; case SNDRV_PCM_TRIGGER_RESUME:

3 4

[alsa-devel] [PATCH] cmi8330: add OPL3 support
by Krzysztof Helt 21 Jan '09

21 Jan '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Add OPL3 handling to the driver and volume control for FM synthesis. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/Kconfig | 1 + sound/isa/cmi8330.c | 30 ++++++++++++++++++++++++++++-- 2 files changed, 29 insertions(+), 2 deletions(-) diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 2d6ce32..ae917e1 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -94,6 +94,7 @@ config SND_CMI8330 tristate "C-Media CMI8330" select SND_WSS_LIB select SND_SB16_DSP + select SND_OPL3_LIB help Say Y here to include support for soundcards based on the C-Media CMI8330 chip. diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index e49aec7..dec6ea5 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -51,6 +51,7 @@ #include <linux/moduleparam.h> #include <sound/core.h> #include <sound/wss.h> +#include <sound/opl3.h> #include <sound/sb.h> #include <sound/initval.h> @@ -79,6 +80,7 @@ static int sbdma16[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; static long wssport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; static int wssirq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; static int wssdma[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; +static long fmport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for CMI8330 soundcard."); @@ -107,6 +109,8 @@ MODULE_PARM_DESC(wssirq, "IRQ # for CMI8330 WSS driver."); module_param_array(wssdma, int, NULL, 0444); MODULE_PARM_DESC(wssdma, "DMA for CMI8330 WSS driver."); +module_param_array(fmport, long, NULL, 0444); +MODULE_PARM_DESC(fmport, "FM port # for CMI8330 driver."); #ifdef CONFIG_PNP static int isa_registered; static int pnp_registered; @@ -219,8 +223,10 @@ WSS_SINGLE("3D Control - Switch", 0, CMI8330_RMUX3D, 5, 1, 1), WSS_SINGLE("PC Speaker Playback Volume", 0, CMI8330_OUTPUTVOL, 3, 3, 0), -WSS_SINGLE("FM Playback Switch", 0, - CMI8330_RECMUX, 3, 1, 1), +WSS_DOUBLE("FM Playback Switch", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), +WSS_DOUBLE("FM Playback Volume", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), WSS_SINGLE(SNDRV_CTL_NAME_IEC958("Input ", CAPTURE, SWITCH), 0, CMI8330_RMUX3D, 7, 1, 1), WSS_SINGLE(SNDRV_CTL_NAME_IEC958("Input ", PLAYBACK, SWITCH), 0, @@ -333,6 +339,7 @@ static int __devinit snd_cmi8330_pnp(int dev, struct snd_cmi8330 *acard, wssport[dev] = pnp_port_start(pdev, 0); wssdma[dev] = pnp_dma(pdev, 0); wssirq[dev] = pnp_irq(pdev, 0); + fmport[dev] = pnp_port_start(pdev, 1); /* allocate SB16 resources */ pdev = acard->play; @@ -487,6 +494,7 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev) { struct snd_cmi8330 *acard; int i, err; + struct snd_opl3 *opl3; acard = card->private_data; err = snd_wss_create(card, wssport[dev] + 4, -1, @@ -530,6 +538,24 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev) snd_printk(KERN_ERR PFX "failed to create pcms\n"); return err; } + if (fmport[dev] != SNDRV_AUTO_PORT) { + if (snd_opl3_create(card, + fmport[dev], fmport[dev] + 2, + OPL3_HW_AUTO, 0, &opl3) < 0) { + snd_printk(KERN_ERR PFX + "no OPL device at 0x%lx-0x%lx ?\n", + fmport[dev], fmport[dev] + 2); + } else { + err = snd_opl3_timer_new(opl3, 0, 1); + if (err < 0) + return err; + + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) + return err; + } + } + strcpy(card->driver, "CMI8330/C3D"); strcpy(card->shortname, "C-Media CMI8330/C3D"); -- 1.5.2.2 ---------------------------------------------------------------------- Zobacz gry internetowe dla dzieci! sprawdz >>>http://link.interia.pl/f202c

3 5

[alsa-devel] aplay hang with 1.0.18
by Will Wagner 21 Jan '09

21 Jan '09

Hi All, I am seeing a problem with aplay hanging when trying to play a wav file. The problem is that it is trying to use an mmap function even though mmap was not initialised. The call stack is: #0 snd_pcm_mmap_begin (pcm=0x400764a0, areas=0x400772c4, offset=0x40061e1c, frames=0x40082f78) at pcm.c:6402 #1 0x40081eb4 in snd_pcm_plugin_write_areas (pcm=0x2b0e8, areas=0xbeb20950, offset=0, size=2048) at pcm_plugin.c:278 #2 0x4006ab20 in snd1_pcm_write_areas (pcm=0x2b0e8, areas=0xbeb20950, offset=0, size=2048, func=0x40081e30 <snd_pcm_plugin_write_areas>) at pcm.c:6654 #3 0x40082578 in snd_pcm_plugin_writei (pcm=0x2b0e8, buffer=0x2b6c8, size=2048) at pcm_plugin.c:361 #4 0x4005e384 in _snd_pcm_writei (pcm=0x21db8, buffer=0x2b6c8, size=2048) at pcm_local.h:516 #5 0x4005e2f8 in snd_pcm_writei (pcm=0x21db8, buffer=0x2b6c8, size=2048) at pcm.c:1243 #6 0x0001045c in pcm_write (data=0x2b6c8 "^", count=2048) at aplay.c:1432 #7 0x000135b8 in playback_go (fd=5, loaded=0, count=424600, rtype=2, name=0xbeb20dd5 "/sdcard/test.wav") at aplay.c:2171 #8 0x00013a0c in playback (name=0xbeb20dd5 "/sdcard/test.wav") at aplay.c:2231 #9 0x0000c38c in main (argc=2, argv=0xbeb20cd4) at aplay.c:641 It gets here even though the mmap_flag was not set. This then causes an infinite loop as no mmap areas have been defined. My setup was working with version 1.0.16. What could have changed to break this? Is it something wrong with my config? Thanks, Will. -- ------------------------------------------------------------------------ Will Wagner will_wagner(a)carallon.com Development Manager Office Tel: +44 (0)20 7371 2032 Carallon Ltd, Studio G20, Shepherds Building, Rockley Rd, London W14 0DA ------------------------------------------------------------------------

2 1

[alsa-devel] [PATCH] sscape: use common MPU401 macros
by Krzysztof Helt 21 Jan '09

21 Jan '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Remove local macros which redefines the common ones. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Just cut few lines from the driver. sound/isa/sscape.c | 12 ++++-------- 1 files changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 6a7f842..681e223 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -89,9 +89,6 @@ MODULE_DEVICE_TABLE(pnp_card, sscape_pnpids); #endif -#define MPU401_IO(i) ((i) + 0) -#define MIDI_DATA_IO(i) ((i) + 0) -#define MIDI_CTRL_IO(i) ((i) + 1) #define HOST_CTRL_IO(i) ((i) + 2) #define HOST_DATA_IO(i) ((i) + 3) #define ODIE_ADDR_IO(i) ((i) + 4) @@ -327,7 +324,7 @@ static int host_write_ctrl_unsafe(unsigned io_base, unsigned char data, */ static inline int verify_mpu401(const struct snd_mpu401 * mpu) { - return ((inb(MIDI_CTRL_IO(mpu->port)) & 0xc0) == 0x80); + return ((inb(MPU401C(mpu)) & 0xc0) == 0x80); } /* @@ -335,7 +332,7 @@ static inline int verify_mpu401(const struct snd_mpu401 * mpu) */ static inline void initialise_mpu401(const struct snd_mpu401 * mpu) { - outb(0, MIDI_DATA_IO(mpu->port)); + outb(0, MPU401D(mpu)); } /* @@ -1191,12 +1188,11 @@ static int __devinit create_sscape(int dev, struct snd_card *card) } #define MIDI_DEVNUM 0 if (sscape->type != SSCAPE_VIVO) { - err = create_mpu401(card, MIDI_DEVNUM, - MPU401_IO(xport), mpu_irq[dev]); + err = create_mpu401(card, MIDI_DEVNUM, xport, mpu_irq[dev]); if (err < 0) { printk(KERN_ERR "sscape: Failed to create " "MPU-401 device at 0x%x\n", - MPU401_IO(xport)); + xport); goto _release_dma; } -- 1.5.2.2 ----------------------------------------------------------------------- Promocja w Speak Up. Kwartal angielskiego za darmo. 3 miesiace nauki gratis. Sprawdz teraz! >> http://link.interia.pl/f2019

1 0

[alsa-devel] [PATCH] hda: 83xxx port 0xe DAC selection
by Matthew Ranostay 21 Jan '09

21 Jan '09

On the 92hd8xxx codecs port 0xe needs the connection selected to be the last DAC in the list. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a4d4afe..7badec0 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -841,10 +841,6 @@ static struct hda_verb stac92hd73xx_10ch_core_init[] = { }; static struct hda_verb stac92hd83xxx_core_init[] = { - /* start of config #1 */ - { 0xe, AC_VERB_SET_CONNECT_SEL, 0x3}, - - /* start of config #2 */ { 0xa, AC_VERB_SET_CONNECT_SEL, 0x0}, { 0xb, AC_VERB_SET_CONNECT_SEL, 0x0}, { 0xd, AC_VERB_SET_CONNECT_SEL, 0x1}, @@ -4754,7 +4750,9 @@ static struct hda_input_mux stac92hd83xxx_dmux = { static int patch_stac92hd83xxx(struct hda_codec *codec) { struct sigmatel_spec *spec; + hda_nid_t conn[STAC92HD83_DAC_COUNT + 1]; int err; + int num_dacs; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -4773,13 +4771,21 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids); spec->multiout.dac_nids = spec->dac_nids; + + /* set port 0xe to select the last DAC + */ + num_dacs = snd_hda_get_connections(codec, 0x0e, + conn, STAC92HD83_DAC_COUNT + 1) - 1; + + snd_hda_codec_write_cache(codec, 0xe, 0, + AC_VERB_SET_CONNECT_SEL, num_dacs); + spec->init = stac92hd83xxx_core_init; switch (codec->vendor_id) { case 0x111d7605: break; default: spec->num_pwrs--; - spec->init++; /* switch to config #2 */ } spec->mixer = stac92hd83xxx_mixer;

2 1

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 21 Jan '09

21 Jan '09

The following changes since commit b2a19d02396c92294abcddee5bd9bd49cc4e4d1c: Mark Brown (1): ASoC: Staticise PCM operations tables are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Ian Molton (1): ASoC: Fix WM9705 capture switch name Mark Brown (1): Merge branch 'for-2.6.29' into for-2.6.30 Timur Tabi (2): ASoC: fix registration of the SoC card in the Freescale MPC8610 drivers ASoC: Allow Freescale MPC8610 audio drivers to be compiled as modules sound/soc/codecs/wm9705.c | 4 ++-- sound/soc/fsl/Kconfig | 16 ++++++++-------- sound/soc/fsl/Makefile | 7 +++++-- sound/soc/fsl/mpc8610_hpcd.c | 18 ++++++------------ 4 files changed, 21 insertions(+), 24 deletions(-)

2 1

[alsa-devel] ASoC update for 2.6.29
by Mark Brown 21 Jan '09

21 Jan '09

The following changes since commit 591046cfc5f1b452db0a4557850ad7f394e38231: Ben Nizette (1): ASoC: atmel_pcm: Remove non-existant header are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.29 Timur Tabi (1): ASoC: fix registration of the SoC card in the Freescale MPC8610 drivers sound/soc/fsl/mpc8610_hpcd.c | 18 ++++++------------ 1 files changed, 6 insertions(+), 12 deletions(-)

2 1

[alsa-devel] [PATCH - snd-powermac 0/1] Driver tree
by Risto Suominen 21 Jan '09

21 Jan '09

Hi Takashi, I still wonder, what about these: --- a/acinclude.m4.org 2008-11-12 12:04:39.000000000 +0200 +++ b/acinclude.m4 2009-01-13 15:04:03.000000000 +0200 @@ -1828,6 +1828,7 @@ CONFIG_SND_TIMER="m" CONFIG_SND_PCM="m" CONFIG_SND_POWERMAC="m" + CONFIG_SND_VMASTER="y" fi fi if alsa_check_kconfig_option "powermac-auto-drc"; then --- a/configure.org 2008-11-12 12:04:44.000000000 +0200 +++ b/configure 2009-01-13 15:04:31.000000000 +0200 @@ -16071,6 +16071,7 @@ CONFIG_SND_TIMER="m" CONFIG_SND_PCM="m" CONFIG_SND_POWERMAC="m" + CONFIG_SND_VMASTER="y" fi fi if alsa_check_kconfig_option "powermac-auto-drc"; then They were needed in making alsa-driver-1.0.18a for the new vmaster controls, but I couldn't see the files in your driver-tree. Risto

2 1

[alsa-devel] [PATCH - snd-powermac 0/5] Some small fixes
by Risto Suominen 21 Jan '09

21 Jan '09

Hi Takashi, What will follow, is two sets of patches, the first for kernel, the second for driver (old kernels). I've run checkpatch.pl and tried to check their patchability against your trees, no functional testing though, not even making. I'll try to send them with git-send-email (something I haven't tried before...). Risto

2 1

[alsa-devel] [PATCH 2.6.29] ASoC: fix registration of the SOC card in the Freescale MPC8610 drivers
by Timur Tabi 20 Jan '09

20 Jan '09

The Freescale MPC8610 driver was defining two SOC card (snd_soc_card) structures, partially initializing each one, but registering only one of them with ASoC. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/mpc8610_hpcd.c | 18 ++++++------------ 1 files changed, 6 insertions(+), 12 deletions(-) diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index bcec3f6..acf39a6 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -183,16 +183,6 @@ static struct snd_soc_ops mpc8610_hpcd_ops = { }; /** - * mpc8610_hpcd_machine: ASoC machine data - */ -static struct snd_soc_card mpc8610_hpcd_machine = { - .probe = mpc8610_hpcd_machine_probe, - .remove = mpc8610_hpcd_machine_remove, - .name = "MPC8610 HPCD", - .num_links = 1, -}; - -/** * mpc8610_hpcd_probe: OF probe function for the fabric driver * * This function gets called when an SSI node is found in the device tree. @@ -455,7 +445,11 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev, machine_data->dai.codec_dai = &cs4270_dai; /* The codec_dai we want */ machine_data->dai.ops = &mpc8610_hpcd_ops; - mpc8610_hpcd_machine.dai_link = &machine_data->dai; + machine_data->machine.probe = mpc8610_hpcd_machine_probe; + machine_data->machine.remove = mpc8610_hpcd_machine_remove; + machine_data->machine.name = "MPC8610 HPCD"; + machine_data->machine.num_links = 1; + machine_data->machine.dai_link = &machine_data->dai; /* Allocate a new audio platform device structure */ sound_device = platform_device_alloc("soc-audio", -1); @@ -465,7 +459,7 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev, goto error; } - machine_data->sound_devdata.card = &mpc8610_hpcd_machine; + machine_data->sound_devdata.card = &machine_data->machine; machine_data->sound_devdata.codec_dev = &soc_codec_device_cs4270; machine_data->machine.platform = &fsl_soc_platform; -- 1.5.5

2 1

[alsa-devel] [PATCH 2.6.29] ASoC: Allow Freescale MPC8610 audio drivers to be compiled as modules
by Timur Tabi 20 Jan '09

20 Jan '09

Change the Kconfig and Makefile options for Freescale MPC8610 audio drivers so that they can be compiled as modules, and simplify the Kconfig choices so that only the platform is selected. Also fix the naming of the driver files to conform to ALSA standards. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/Kconfig | 16 ++++++++-------- sound/soc/fsl/Makefile | 7 +++++-- 2 files changed, 13 insertions(+), 10 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 95c12b2..bd6974e 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,17 +1,17 @@ config SND_SOC_OF_SIMPLE tristate +# ASoC platform support for the Freescale MPC8610 SOC. This compiles drivers +# for the SSI and the Elo DMA controller. You will still need to select +# a platform driver and a codec driver. config SND_SOC_MPC8610 - bool "ALSA SoC support for the MPC8610 SOC" - depends on MPC8610_HPCD - default y if MPC8610 - help - Say Y if you want to add support for codecs attached to the SSI - device on an MPC8610. + tristate + depends on MPC8610 config SND_SOC_MPC8610_HPCD - bool "ALSA SoC support for the Freescale MPC8610 HPCD board" - depends on SND_SOC_MPC8610 + tristate "ALSA SoC support for the Freescale MPC8610 HPCD board" + depends on SND_SOC && MPC8610_HPCD + select SND_SOC_MPC8610 select SND_SOC_CS4270 select SND_SOC_CS4270_VD33_ERRATA default y if MPC8610_HPCD diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 035da4a..f85134c 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -2,10 +2,13 @@ obj-$(CONFIG_SND_SOC_OF_SIMPLE) += soc-of-simple.o # MPC8610 HPCD Machine Support -obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += mpc8610_hpcd.o +snd-soc-mpc8610-hpcd-objs := mpc8610_hpcd.o +obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += snd-soc-mpc8610-hpcd.o # MPC8610 Platform Support -obj-$(CONFIG_SND_SOC_MPC8610) += fsl_ssi.o fsl_dma.o +snd-soc-fsl-ssi-objs := fsl_ssi.o +snd-soc-fsl-dma-objs := fsl_dma.o +obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o -- 1.5.5

3 3

[alsa-devel] [PATCH - snd-powermac 1/1] Allow input from mic in iBook G3 Dual-USB (older kernels)
by Risto Suominen 20 Jan '09

20 Jan '09

Allow input from microphone on iBook G3 Dual-USB (Tumbler) on older kernels. Signed-off-by: Risto Suominen <Risto.Suominen(a)gmail.com> diff --git a/ppc/pmac_old.c b/ppc/pmac_old.c index 1bb65fe..c6ebc11 100644 --- a/ppc/pmac_old.c +++ b/ppc/pmac_old.c @@ -964,7 +964,8 @@ static int __init snd_pmac_detect(struct snd_pmac *chip) } if (device_is_compatible(sound, "tumbler")) { chip->model = PMAC_TUMBLER; - chip->can_capture = machine_is_compatible("PowerMac4,2"); + chip->can_capture = machine_is_compatible("PowerMac4,2") + || machine_is_compatible("PowerBook4,1"); chip->can_duplex = 0; // chip->can_byte_swap = 0; /* FIXME: check this */ chip->num_freqs = ARRAY_SIZE(tumbler_freqs); -- 1.5.4.3

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[alsa-devel] [PATCH - snd-powermac 1/5] Correct HP detection and input selectors for PMac 5500
by Risto Suominen 20 Jan '09

20 Jan '09

Correct headphone detection and input selectors for PowerMac 5500 (AWACS). Signed-off-by: Risto Suominen <Risto.Suominen(a)gmail.com> diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c index 7bd33e6..0258ccb 100644 --- a/sound/ppc/awacs.c +++ b/sound/ppc/awacs.c @@ -767,6 +767,7 @@ static void snd_pmac_awacs_resume(struct snd_pmac *chip) #endif /* CONFIG_PM */ #define IS_PM7500 (machine_is_compatible("AAPL,7500")) +#define IS_PM5500 (machine_is_compatible("AAPL,e411")) #define IS_BEIGE (machine_is_compatible("AAPL,Gossamer")) #define IS_IMAC1 (machine_is_compatible("PowerMac2,1")) #define IS_IMAC2 (machine_is_compatible("PowerMac2,2") \ @@ -858,6 +859,7 @@ int __init snd_pmac_awacs_init(struct snd_pmac *chip) { int pm7500 = IS_PM7500; + int pm5500 = IS_PM5500; int beige = IS_BEIGE; int g4agp = IS_G4AGP; int imac; @@ -915,7 +917,7 @@ snd_pmac_awacs_init(struct snd_pmac *chip) /* set headphone-jack detection bit */ switch (chip->model) { case PMAC_AWACS: - chip->hp_stat_mask = pm7500 ? MASK_HDPCONN + chip->hp_stat_mask = pm7500 || pm5500 ? MASK_HDPCONN : MASK_LOCONN; break; case PMAC_SCREAMER: @@ -954,7 +956,7 @@ snd_pmac_awacs_init(struct snd_pmac *chip) return err; if (beige || g4agp) ; - else if (chip->model == PMAC_SCREAMER) + else if (chip->model == PMAC_SCREAMER || pm5500) err = build_mixers(chip, ARRAY_SIZE(snd_pmac_screamer_mixers2), snd_pmac_screamer_mixers2); else if (!pm7500) -- 1.5.4.3

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[alsa-devel] Any CMedia CMI8788 "Oxygen" and ASUS Xonar D1 "virtuoso" documentation available?
by John L. Utz III 20 Jan '09

20 Jan '09

Hi; I just tried to find the docs for the CMI8788 on www.cmedia.com.tw and i was not successful. Nor have i been able to find any virtuoso docs at www.asus.com. I know that Clemens wrote the driver, but I dont know if it was written from scratch with docs or ported from a 4front driver. currently, there are some bugs with channel mapping on the rear channel in Surround40 mode that i would like to fix tnx for any info you folks can provide johnu

2 2

[alsa-devel] ICE1712 softdevices fixes
by Diego E. 'Flameeyes' Pettenò 20 Jan '09

20 Jan '09

The following patches fixes the default device as they appear for ICE1712-based cards. Without this cards, 'aplay -L' will show a series of devices like surround40 and surround71 that are not meaningful for this card, and a front device that would not work.

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[alsa-devel] Could we get the card name in /sys/clas/sound/card0/name?
by Lennart Poettering 20 Jan '09

20 Jan '09

Heya! Hmm, it would be nifty if we could get the alsa card name (i.e. the one shown in [] in /proc/pcm/cards) as an attribute file in /sys/class/sound/cardX/name. Even more useful would be if this property would be writable so that the cards may be renamed during runtime. This could be very useful to establish stable card names via udev rules, similar to what is already done to network interface names ("eth0", ...). Would such a change be acceptable? Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

2 1

[alsa-devel] GIT branch for ICE1712-related fixes
by flameeyes＠gmail.com 20 Jan '09

20 Jan '09

At git://git.flameeyes.eu/others/alsa/alsa-lib.git (branch ice1712) you can find some fixes for ICE1712-based cards: Diego E. 'Flameeyes' Pettenò (2): Remove definition for surround sound devices on ICE1712. Make the front:idx device work for ICE1712. With this aplay -L only shows devices that actually do work, rather than a series of devices where just iec958 works. HTH, -- Diego "Flameeyes" Pettenò http://blog.flameeyes.eu/

2 1

[alsa-devel] [PATCH] hda: default dmuxes to first entry in dinput_mux
by Matthew Ranostay 20 Jan '09

20 Jan '09

When using a custom spec->dinput_mux sometimes the value selected is incorrect due to not have a previously selected default. To remedy this patch selects the first items index in spec->dinput_mux and sets all the dmuxes thusly to it. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index e2b673a..3f5dc1b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3663,7 +3663,17 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out spec->mixers[spec->num_mixers++] = spec->kctls.list; spec->input_mux = &spec->private_imux; - if (!spec->dinput_mux) + if (spec->dinput_mux) { + int idx = spec->dinput_mux->items[0].index; + int i; + + /* sometimes using a custom dinput_mux causes the incorrect + * value to selected on startup, but sure a valid one is + * selected by default */ + for (i = 0; i < spec->num_dmuxes; i++) + snd_hda_codec_write_cache(codec, spec->dmux_nids[i], 0, + AC_VERB_SET_CONNECT_SEL, idx); + } else spec->dinput_mux = &spec->private_dimux; spec->sinput_mux = &spec->private_smux; spec->mono_mux = &spec->private_mono_mux;

2 1

[alsa-devel] Firmware
by soundprizm＠gmail.com 20 Jan '09

20 Jan '09

Are any plans for the development of firmware for the Tascam us-1641? If not, can anybody point me in the right direction in writing or obtaining this? Any feedback is much appreciated! Thanks, Donnie Menig.

1 0

[alsa-devel] "softswitch" feature?
by Nicolas Boullis 20 Jan '09

20 Jan '09

Hello, I'd like to implement a PCM device with 2 slaves, that sends its stream to either slave according to a user-defined switch. I need this for a card that is able to play either 2-channel 16-bit big-endian frames to its analog output or IEC958 frames to its digital output. I'd like to setup a "default" device for this card that send sound to either output according to a switch. Is such a thing feasible with the existing plugins? If not, do you think it would be easy to implement? Note that, although my card implements this as 2 separate devices, they can't be open simultaneously since both use the same hardware fifo. Hence, I think it would require that the unused slave not be open. I also can imagine othen applications for such a feature, such as with a VOIP application to use either a standard soundcard connected to loudspeakers and a microphone, or a bluetooth headset. Reagrds, Nicolas PS: FWIW, the card I'm talking about is a hardware MPEG decoder board, whose driver I'm the maintainer of, an which implements an ALSA driver for its audio part. It's current card-specific configuration is available at http://freehg.org/u/nboullis/em8300/file/tip/modules/EM8300.conf All comments are welcome.

1 0

[alsa-devel] {Q}simple selection of analog ports other than the headphones
by S. Aguinaga 19 Jan '09

19 Jan '09

Dear Colleagues, I have installed a fresh Fedora 9, and I have an Echo Layla3G (pci) card. All I had to do was install the firmware, and it to playback out of Layla headphones. This device has other Outputs, how I can I easily select the other channels for playback? aplay -l show these (along withthe Intel HDA for the built-in audio device): card 1: Layla3G [Layla3G], device 0: Analog PCM [Layla3G] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 so is it possible to type: aplay -D hw:1,0 thewavefile.wav and specify the subdevice? Or do I need to specify in my .asoundrc file an alias? with -D hw:1,0 I get sound out of Layla's headphone port only. If you have links to useful howtos or documentation, please let me know. // Sal Aguinaga // Northwestern University

1 0

[alsa-devel] [Patch] WM9705 tweak
by Ian Molton 19 Jan '09

19 Jan '09

Hi folks, Just a little patch to make recording work properly on wm9705 (the mute is hidden on alsamixer otherwise. (that and the control was poorly named). -Ian From 47fa5f6a17ff9a7e546c7fae84b0a9c0b97bc74d Mon Sep 17 00:00:00 2001 From: Ian Molton <ian(a)mnementh.co.uk> Date: Mon, 19 Jan 2009 12:46:54 +0000 Subject: [PATCH] ASoC: codec: WM9705 Fix capture switch name This patch fixes the acpture switch name so that it shows up in alsamixer. Without this, its not possible to unmute the capture PGA. Signed-off-by: Ian Molton <iann(a)mnementh.co.uk> --- sound/soc/codecs/wm9705.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index cb26b6a..01bfd5d 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -57,8 +57,8 @@ static const struct snd_kcontrol_new wm9705_snd_ac97_controls[] = { SOC_DOUBLE("CD Playback Volume", AC97_CD, 8, 0, 31, 1), SOC_SINGLE("Mic Playback Volume", AC97_MIC, 0, 31, 1), SOC_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 6, 1, 0), - SOC_DOUBLE("PCM Capture Volume", AC97_REC_GAIN, 8, 0, 15, 0), - SOC_SINGLE("PCM Capture Switch", AC97_REC_GAIN, 15, 1, 1), + SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 15, 0), + SOC_SINGLE("Capture ADC Switch", AC97_REC_GAIN, 15, 1, 1), }; static const char *wm9705_mic[] = {"Mic 1", "Mic 2"}; -- 1.5.6.5

2 5

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 19 Jan '09

19 Jan '09

The following changes since commit 2905a5df4a1600bc3c9ecace2fa65404196d1fda: Mark Brown (1): Merge branch 'for-2.6.29' into for-2.6.30 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Ian Molton (1): ASoC: machine support for Toshiba e740 PDA Mark Brown (4): ASoC: Ignore output frequency for WM9713 PLL ASoC: Implement support for CLK_POUT as MCLK on Zylonite ASoC: fixes to caching implementations ASoC: Staticise PCM operations tables arch/arm/mach-pxa/e740.c | 5 + arch/arm/mach-pxa/include/mach/eseries-gpio.h | 5 + sound/soc/atmel/atmel-pcm.c | 2 +- sound/soc/au1x/dbdma2.c | 2 +- sound/soc/blackfin/bf5xx-ac97-pcm.c | 2 +- sound/soc/blackfin/bf5xx-i2s-pcm.c | 2 +- sound/soc/codecs/ac97.c | 2 - sound/soc/codecs/ad1980.c | 4 +- sound/soc/codecs/twl4030.c | 3 + sound/soc/codecs/wm8580.c | 4 +- sound/soc/codecs/wm8728.c | 4 +- sound/soc/codecs/wm8753.c | 4 +- sound/soc/codecs/wm8990.c | 4 +- sound/soc/codecs/wm9712.c | 4 +- sound/soc/codecs/wm9713.c | 14 +- sound/soc/davinci/davinci-pcm.c | 2 +- sound/soc/omap/omap-pcm.c | 2 +- sound/soc/pxa/Kconfig | 9 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/e740_wm9705.c | 213 +++++++++++++++++++++++++ sound/soc/pxa/zylonite.c | 101 ++++++++++-- 21 files changed, 345 insertions(+), 45 deletions(-) create mode 100644 sound/soc/pxa/e740_wm9705.c

2 4

[alsa-devel] Any Docs for Creative X-Fi USB
by Mandar Joshi 19 Jan '09

19 Jan '09

Hello, Are there any technical documents available for Creative X-Fi USB that'll help write a driver? I couldn't find any on http://connect.creativelabs.com Regards Mandar Joshi

1 0

[alsa-devel] Cross compile issues with alsa-utils
by Will Wagner 19 Jan '09

19 Jan '09

Hi All, I have been cross compiling alsa-utils with buildroot (www.buildroot.org) and have run into two issues. The first is that alsactl is using index & rindex which are legacy functions not supported with all cross compile toolchains. Attached is a patch to replace the calls with strchr and strrchr. The second issue is again with alsactl. It is trying to generate the alsactl_init.7 man page but I do not have xmlto to convert it to a man page. Could a configure option be added to not build any docs - similar to alsa-lib configure. Thanks, Will. -- ------------------------------------------------------------------------ Will Wagner will_wagner(a)carallon.com Development Manager Office Tel: +44 (0)20 7371 2032 Carallon Ltd, Studio G20, Shepherds Building, Rockley Rd, London W14 0DA ------------------------------------------------------------------------ --- a/alsactl/init_parse.c 2009-01-15 10:44:48.000000000 +0000 +++ b/alsactl/init_parse.c 2009-01-15 10:46:41.000000000 +0000 @@ -381,7 +381,7 @@ static int set_ctl_value(struct space *s snd_ctl_elem_value_set_integer(space->ctl_value, idx, val); } else if (items > 2 && value[items-2] == 'd' && value[items-1] == 'B') { val = strtol(value, NULL, 0) * 100; - if ((pos2 = index(value, '.')) != NULL) { + if ((pos2 = strchr(value, '.')) != NULL) { if (isdigit(*(pos2-1)) && isdigit(*(pos2-2))) { if (val < 0) val -= strtol(pos2 + 1, NULL, 0); @@ -1253,7 +1253,7 @@ static char *new_root_dir(const char *fi res = strdup(filename); if (res) { - tmp = rindex(res, '/'); + tmp = strrchr(res, '/'); if (tmp) *tmp = '\0'; }

2 1

[alsa-devel] [PATCH] alsa-info: Use pgrep with -f for matching against the process' cmdline.
by Ozan Çağlayan 19 Jan '09

19 Jan '09

Latest alsa-info.sh has a sound daemon detection snippet which uses pgrep to find out if any sound daemon is loaded or not. The full paths to the binaries are given to pgrep but pgrep matches its first parameter against the "process name" by default. Call pgrep with -f to match against the process' command line to fix the detection code. Signed-off-by: Ozan Çağlayan <ozan(a)pardus.org.tr> --- utils/alsa-info.sh | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/utils/alsa-info.sh b/utils/alsa-info.sh index 1be147f..be10362 100755 --- a/utils/alsa-info.sh +++ b/utils/alsa-info.sh @@ -375,21 +375,21 @@ echo "!!Sound Servers on this system" >> $FILE echo "!!----------------------------" >> $FILE echo "" >> $FILE if [[ -n $PAINST ]];then -[[ `pgrep $PAINST` ]] && PARUNNING="Yes" || PARUNNING="No" +[[ `pgrep -f $PAINST` ]] && PARUNNING="Yes" || PARUNNING="No" echo "Pulseaudio:" >> $FILE echo " Installed - Yes ($PAINST)" >> $FILE echo " Running - $PARUNNING" >> $FILE echo "" >> $FILE fi if [[ -n $ESDINST ]];then -[[ `pgrep $ESDINST` ]] && ESDRUNNING="Yes" || ESDRUNNING="No" +[[ `pgrep -f $ESDINST` ]] && ESDRUNNING="Yes" || ESDRUNNING="No" echo "ESound Daemon:" >> $FILE echo " Installed - Yes ($ESDINST)" >> $FILE echo " Running - $ESDRUNNING" >> $FILE echo "" >> $FILE fi if [[ -n $ARTSINST ]];then -[[ `pgrep $ARTSINST` ]] && ARTSRUNNING="Yes" || ARTSRUNNING="No" +[[ `pgrep -f $ARTSINST` ]] && ARTSRUNNING="Yes" || ARTSRUNNING="No" echo "aRts:" >> $FILE echo " Installed - Yes ($ARTSINST)" >> $FILE echo " Running - $ARTSRUNNING" >> $FILE -- 1.6.1

1 0

[alsa-devel] ALSA 1.0.19 release
by Jaroslav Kysela 19 Jan '09

19 Jan '09

Hello, I released ALSA 1.0.19 packages (except alsa-oss and alsa-python - no changes). Description of changes is available at URL bellow: http://www.alsa-project.org/main/index.php/Changes_v1.0.18_v1.0.19 Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

1 0

[alsa-devel] [PATCH] hda: fix invalid power mapping masks
by Matthew Ranostay 18 Jan '09

18 Jan '09

Fixed invalid power mappings for ports 0xd and 0xe on 93hd83xxx codecs. They were shifted right one too many bits. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index faef1ca..a4d4afe 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -334,7 +334,7 @@ static hda_nid_t stac92hd83xxx_slave_dig_outs[2] = { }; static unsigned int stac92hd83xxx_pwr_mapping[4] = { - 0x03, 0x0c, 0x10, 0x40, + 0x03, 0x0c, 0x20, 0x80, }; static hda_nid_t stac92hd83xxx_amp_nids[1] = {

2 1

[alsa-devel] hda_codec: Unknown model for ALC268
by Stefanie 17 Jan '09

17 Jan '09

Hello! I have problems with the sound chip of my notebook. It is a Medion akoya WIM 2220. Actually, there is a driver for the chip, but somehow, I can't get it to work. I'm desperate and don't know what to do. My friend helped my to gather the following information: This is from lspci: 00:1b.0 0403: 8086:284b (rev 04) 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 04) Subsystem: Wistron Corp. Device 4085 Flags: fast devsel, IRQ 22 Memory at fa200000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Count=1/1 Enable- Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 Capabilities: [100] Virtual Channel <?> Capabilities: [130] Root Complex Link <?> Kernel modules: snd-hda-intelk <?> I have found this in my syslog while trying to load snd-hda-intel: Jan 10 21:30:16 Mescalin HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 Jan 10 21:30:16 Mescalin HDA Intel 0000:00:1b.0: setting latency timer to 64 Jan 10 21:30:16 Mescalin hda_codec: Unknown model for ALC268, trying auto-probe from BIOS... Jan 10 21:30:16 Mescalin hda_codec: Cannot set up configuration from BIOS. Using base mode... Jan 10 21:30:16 Mescalin SPDIF already defined Jan 10 21:30:16 Mescalin HDA Intel 0000:00:1b.0: PCI INT A disabled Jan 10 21:30:16 Mescalin HDA Intel: probe of 0000:00:1b.0 failed with error -16 Thanks in advance for your help. Stefanie

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[alsa-devel] [Pull request] Support for wm9705 codec and two machines that use it.
by Ian Molton 17 Jan '09

17 Jan '09

The following changes since commit 6d3455ecb7d564789e8dd493fb876f4a8706e3b2: Mark Brown (1): Fix leftover merge issue. are available in the git repository at: git://git.mnementh.co.uk/linux-2.6-im.git asoc Ian Molton (3): ASoc: codec: Driver for wm9705 AC97 codec. ASoC: machine driver for Toshiba e750 ASoC: machine driver for Toshiba e800 arch/arm/mach-pxa/e750.c | 5 + arch/arm/mach-pxa/include/mach/eseries-gpio.h | 10 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm9705.c | 401 ++++++++++++++++++ sound/soc/codecs/wm9705.h | 14 + sound/soc/pxa/Kconfig | 9 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/e750_wm9705.c | 185 ++++++++++++ sound/soc/pxa/e800_wm9712.c | 115 +++++++- 10 files changed, 733 insertions(+), 14 deletions(-) create mode 100644 sound/soc/codecs/wm9705.c create mode 100644 sound/soc/codecs/wm9705.h create mode 100644 sound/soc/pxa/e750_wm9705.c

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[alsa-devel] [Pull request] ASoC: machine support for Toshiba e740 PDA
by Ian Molton 17 Jan '09

17 Jan '09

Hi folks, This is complete support for the wm9705 codec on the e740. Everything works. Please review. Patch attached below the pull request. Mark, if you want the pull request minus the other patchset in my tree (caching stuff) then I can redo. Otherwise, feel free to cherrypick. Thanks, -Ian The following changes since commit 7ff27c9c1f7a7b7c86d3138060b1b406ad5dd6af: Ian Molton (1): ASoC: fixes to caching implementations are available in the git repository at: git://git.mnementh.co.uk/linux-2.6-im.git asoc-pu Ian Molton (1): ASoC: machine support for Toshiba e740 PDA arch/arm/mach-pxa/e740.c | 5 + arch/arm/mach-pxa/include/mach/eseries-gpio.h | 5 + sound/soc/pxa/Kconfig | 9 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/e740_wm9705.c | 213 +++++++++++++++++++++++++ 5 files changed, 234 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/e740_wm9705.c From 45bf08e654a94cfc8933e57b7c275cf565783fe3 Mon Sep 17 00:00:00 2001 From: Ian Molton <ian(a)mnementh.co.uk> Date: Sat, 17 Jan 2009 15:11:06 +0000 Subject: [PATCH] ASoC: machine support for Toshiba e740 PDA This patch provides suupport for the wm9705 AC97 codec on the Toshiba e740. Note: The e740 has a hard headphone switch that turns the speaker off and is not software detectable or controlable. Also both headphone and speaker amps share a common output enable. Signed-off-by: Ian Molton <ian(a)mnementh.co.uk> --- arch/arm/mach-pxa/e740.c | 5 + arch/arm/mach-pxa/include/mach/eseries-gpio.h | 5 + sound/soc/pxa/Kconfig | 9 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/e740_wm9705.c | 213 +++++++++++++++++++++++++ 5 files changed, 234 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/e740_wm9705.c diff --git a/arch/arm/mach-pxa/e740.c b/arch/arm/mach-pxa/e740.c index 6d48e00..a6fff78 100644 --- a/arch/arm/mach-pxa/e740.c +++ b/arch/arm/mach-pxa/e740.c @@ -135,6 +135,11 @@ static unsigned long e740_pin_config[] __initdata = { /* IrDA */ GPIO38_GPIO | MFP_LPM_DRIVE_HIGH, + /* Audio power control */ + GPIO16_GPIO, /* AC97 codec AVDD2 supply (analogue power) */ + GPIO40_GPIO, /* Mic amp power */ + GPIO41_GPIO, /* Headphone amp power */ + /* PC Card */ GPIO8_GPIO, /* CD0 */ GPIO44_GPIO, /* CD1 */ diff --git a/arch/arm/mach-pxa/include/mach/eseries-gpio.h b/arch/arm/mach-pxa/include/mach/eseries-gpio.h index 6d6e4d8..f3e5509 100644 --- a/arch/arm/mach-pxa/include/mach/eseries-gpio.h +++ b/arch/arm/mach-pxa/include/mach/eseries-gpio.h @@ -45,6 +45,11 @@ /* e7xx IrDA power control */ #define GPIO_E7XX_IR_OFF 38 +/* e740 audio control GPIOs */ +#define GPIO_E740_WM9705_nAVDD2 16 +#define GPIO_E740_MIC_ON 40 +#define GPIO_E740_AMP_ON 41 + /* e750 audio control GPIOs */ #define GPIO_E750_HP_AMP_OFF 4 #define GPIO_E750_SPK_AMP_OFF 7 diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index b9b1a3f..958ac3f 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -61,6 +61,15 @@ config SND_PXA2XX_SOC_TOSA Say Y if you want to add support for SoC audio on Sharp Zaurus SL-C6000x models (Tosa). +config SND_PXA2XX_SOC_E740 + tristate "SoC AC97 Audio support for e740" + depends on SND_PXA2XX_SOC && MACH_E740 + select SND_SOC_WM9705 + select SND_PXA2XX_SOC_AC97 + help + Say Y if you want to add support for SoC audio on the + toshiba e740 PDA + config SND_PXA2XX_SOC_E750 tristate "SoC AC97 Audio support for e750" depends on SND_PXA2XX_SOC && MACH_E750 diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index c7d4cce..97a51a8 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -13,6 +13,7 @@ obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o snd-soc-corgi-objs := corgi.o snd-soc-poodle-objs := poodle.o snd-soc-tosa-objs := tosa.o +snd-soc-e740-objs := e740_wm9705.o snd-soc-e750-objs := e750_wm9705.o snd-soc-e800-objs := e800_wm9712.o snd-soc-spitz-objs := spitz.o @@ -23,6 +24,7 @@ snd-soc-zylonite-objs := zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o +obj-$(CONFIG_SND_PXA2XX_SOC_E740) += snd-soc-e740.o obj-$(CONFIG_SND_PXA2XX_SOC_E750) += snd-soc-e750.o obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c new file mode 100644 index 0000000..ac36176 --- /dev/null +++ b/sound/soc/pxa/e740_wm9705.c @@ -0,0 +1,213 @@ +/* + * e740-wm9705.c -- SoC audio for e740 + * + * Copyright 2007 (c) Ian Molton <spyro(a)f2s.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; version 2 ONLY. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/gpio.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/pxa-regs.h> +#include <mach/hardware.h> +#include <mach/audio.h> +#include <mach/eseries-gpio.h> + +#include <asm/mach-types.h> + +#include "../codecs/wm9705.h" +#include "pxa2xx-pcm.h" +#include "pxa2xx-ac97.h" + + +#define E740_AUDIO_OUT 1 +#define E740_AUDIO_IN 2 + +static int e740_audio_power; + +static void e740_sync_audio_power(int status) +{ + gpio_set_value(GPIO_E740_WM9705_nAVDD2, !status); + gpio_set_value(GPIO_E740_AMP_ON, (status & E740_AUDIO_OUT) ? 1 : 0); + gpio_set_value(GPIO_E740_MIC_ON, (status & E740_AUDIO_IN) ? 1 : 0); +} + +static int e740_mic_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (event & SND_SOC_DAPM_PRE_PMU) + e740_audio_power |= E740_AUDIO_IN; + else if (event & SND_SOC_DAPM_POST_PMD) + e740_audio_power &= ~E740_AUDIO_IN; + + e740_sync_audio_power(e740_audio_power); + + return 0; +} + +static int e740_output_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (event & SND_SOC_DAPM_PRE_PMU) + e740_audio_power |= E740_AUDIO_OUT; + else if (event & SND_SOC_DAPM_POST_PMD) + e740_audio_power &= ~E740_AUDIO_OUT; + + e740_sync_audio_power(e740_audio_power); + + return 0; +} + +static const struct snd_soc_dapm_widget e740_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic (Internal)", NULL), + SND_SOC_DAPM_PGA_E("Output Amp", SND_SOC_NOPM, 0, 0, NULL, 0, + e740_output_amp_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("Mic Amp", SND_SOC_NOPM, 0, 0, NULL, 0, + e740_mic_amp_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Output Amp", NULL, "LOUT"}, + {"Output Amp", NULL, "ROUT"}, + {"Output Amp", NULL, "MONOOUT"}, + + {"Speaker", NULL, "Output Amp"}, + {"Headphone Jack", NULL, "Output Amp"}, + + {"MIC1", NULL, "Mic Amp"}, + {"Mic Amp", NULL, "Mic (Internal)"}, +}; + +static int e740_ac97_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_nc_pin(codec, "HPOUTL"); + snd_soc_dapm_nc_pin(codec, "HPOUTR"); + snd_soc_dapm_nc_pin(codec, "PHONE"); + snd_soc_dapm_nc_pin(codec, "LINEINL"); + snd_soc_dapm_nc_pin(codec, "LINEINR"); + snd_soc_dapm_nc_pin(codec, "CDINL"); + snd_soc_dapm_nc_pin(codec, "CDINR"); + snd_soc_dapm_nc_pin(codec, "PCBEEP"); + snd_soc_dapm_nc_pin(codec, "MIC2"); + + snd_soc_dapm_new_controls(codec, e740_dapm_widgets, + ARRAY_SIZE(e740_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link e740_dai[] = { + { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], + .codec_dai = &wm9705_dai[WM9705_DAI_AC97_HIFI], + .init = e740_ac97_init, + }, + { + .name = "AC97 Aux", + .stream_name = "AC97 Aux", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], + .codec_dai = &wm9705_dai[WM9705_DAI_AC97_AUX], + }, +}; + +static struct snd_soc_card e740 = { + .name = "Toshiba e740", + .platform = &pxa2xx_soc_platform, + .dai_link = e740_dai, + .num_links = ARRAY_SIZE(e740_dai), +}; + +static struct snd_soc_device e740_snd_devdata = { + .card = &e740, + .codec_dev = &soc_codec_dev_wm9705, +}; + +static struct platform_device *e740_snd_device; + +static int __init e740_init(void) +{ + int ret; + + if (!machine_is_e740()) + return -ENODEV; + + ret = gpio_request(GPIO_E740_MIC_ON, "Mic amp"); + if (ret) + return ret; + + ret = gpio_request(GPIO_E740_AMP_ON, "Output amp"); + if (ret) + goto free_mic_amp_gpio; + + ret = gpio_request(GPIO_E740_WM9705_nAVDD2, "Audio power"); + if (ret) + goto free_op_amp_gpio; + + /* Disable audio */ + ret = gpio_direction_output(GPIO_E740_MIC_ON, 0); + if (ret) + goto free_apwr_gpio; + ret = gpio_direction_output(GPIO_E740_AMP_ON, 0); + if (ret) + goto free_apwr_gpio; + ret = gpio_direction_output(GPIO_E740_WM9705_nAVDD2, 1); + if (ret) + goto free_apwr_gpio; + + e740_snd_device = platform_device_alloc("soc-audio", -1); + if (!e740_snd_device) { + ret = -ENOMEM; + goto free_apwr_gpio; + } + + platform_set_drvdata(e740_snd_device, &e740_snd_devdata); + e740_snd_devdata.dev = &e740_snd_device->dev; + ret = platform_device_add(e740_snd_device); + + if (!ret) + return 0; + +/* Fail gracefully */ + platform_device_put(e740_snd_device); +free_apwr_gpio: + gpio_free(GPIO_E740_WM9705_nAVDD2); +free_op_amp_gpio: + gpio_free(GPIO_E740_AMP_ON); +free_mic_amp_gpio: + gpio_free(GPIO_E740_MIC_ON); + + return ret; +} + +static void __exit e740_exit(void) +{ + platform_device_unregister(e740_snd_device); +} + +module_init(e740_init); +module_exit(e740_exit); + +/* Module information */ +MODULE_AUTHOR("Ian Molton <spyro(a)f2s.com>"); +MODULE_DESCRIPTION("ALSA SoC driver for e740"); +MODULE_LICENSE("GPL v2"); -- 1.5.6.5

2 1

[alsa-devel] [PATCH] sscape: fix incorrect timeout after microcode upload
by Krzysztof Helt 17 Jan '09

17 Jan '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> A comment states that one should wait up to 5 secs while a waiting loop waits only 5 system ticks. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Ted tested that patch and it allowed him uploading a microcode into the Soundscape Elite. However, midi does not work for him yet. sound/isa/sscape.c | 20 ++++++++++---------- 1 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index bc44916..347a880 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -393,19 +393,19 @@ static int sscape_wait_dma_unsafe(unsigned io_base, enum GA_REG reg, unsigned ti */ static int obp_startup_ack(struct soundscape *s, unsigned timeout) { - while (timeout != 0) { + unsigned long end_time = jiffies + msecs_to_jiffies(timeout); + + while (time_before(jiffies, end_time)) { unsigned long flags; unsigned char x; - schedule_timeout_uninterruptible(1); - spin_lock_irqsave(&s->lock, flags); x = inb(HOST_DATA_IO(s->io_base)); spin_unlock_irqrestore(&s->lock, flags); if ((x & 0xfe) == 0xfe) return 1; - --timeout; + msleep(10); } /* while */ return 0; @@ -420,19 +420,19 @@ static int obp_startup_ack(struct soundscape *s, unsigned timeout) */ static int host_startup_ack(struct soundscape *s, unsigned timeout) { - while (timeout != 0) { + unsigned long end_time = jiffies + msecs_to_jiffies(timeout); + + while (time_before(jiffies, end_time)) { unsigned long flags; unsigned char x; - schedule_timeout_uninterruptible(1); - spin_lock_irqsave(&s->lock, flags); x = inb(HOST_DATA_IO(s->io_base)); spin_unlock_irqrestore(&s->lock, flags); if (x == 0xfe) return 1; - --timeout; + msleep(10); } /* while */ return 0; @@ -529,10 +529,10 @@ static int upload_dma_data(struct soundscape *s, * give it 5 seconds (max) ... */ ret = 0; - if (!obp_startup_ack(s, 5)) { + if (!obp_startup_ack(s, 5000)) { snd_printk(KERN_ERR "sscape: No response from on-board processor after upload\n"); ret = -EAGAIN; - } else if (!host_startup_ack(s, 5)) { + } else if (!host_startup_ack(s, 5000)) { snd_printk(KERN_ERR "sscape: SoundScape failed to initialise\n"); ret = -EAGAIN; } -- 1.5.2.2 ----------------------------------------------------------------------- Promocja w Speak Up. Kwartal angielskiego za darmo. 3 miesiace nauki gratis. Sprawdz teraz! >> http://link.interia.pl/f2019

3 7

[alsa-devel] [PATCH] wss-lib: remove "pops" before each played sound
by Krzysztof Helt 17 Jan '09

17 Jan '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> A WSS codec is autocalibrated each time before playing sound. Do only one calibration during codec initialization. Complete snd_wss_calibrate_mute to mute loopback volume as well. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/wss/wss_lib.c | 41 +++++++++++++---------------------------- 1 files changed, 13 insertions(+), 28 deletions(-) diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 13299ae..f0c0be5 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -181,25 +181,6 @@ static void snd_wss_wait(struct snd_wss *chip) udelay(100); } -static void snd_wss_outm(struct snd_wss *chip, unsigned char reg, - unsigned char mask, unsigned char value) -{ - unsigned char tmp = (chip->image[reg] & mask) | value; - - snd_wss_wait(chip); -#ifdef CONFIG_SND_DEBUG - if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) - snd_printk("outm: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value); -#endif - chip->image[reg] = tmp; - if (!chip->calibrate_mute) { - wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); - wmb(); - wss_outb(chip, CS4231P(REG), tmp); - mb(); - } -} - static void snd_wss_dout(struct snd_wss *chip, unsigned char reg, unsigned char value) { @@ -587,7 +568,15 @@ static void snd_wss_calibrate_mute(struct snd_wss *chip, int mute) chip->image[CS4231_RIGHT_INPUT]); snd_wss_dout(chip, CS4231_LOOPBACK, chip->image[CS4231_LOOPBACK]); + } else { + snd_wss_dout(chip, CS4231_LEFT_INPUT, + 0); + snd_wss_dout(chip, CS4231_RIGHT_INPUT, + 0); + snd_wss_dout(chip, CS4231_LOOPBACK, + 0xfd); } + snd_wss_dout(chip, CS4231_AUX1_LEFT_INPUT, mute | chip->image[CS4231_AUX1_LEFT_INPUT]); snd_wss_dout(chip, CS4231_AUX1_RIGHT_INPUT, @@ -630,7 +619,6 @@ static void snd_wss_playback_format(struct snd_wss *chip, int full_calib = 1; mutex_lock(&chip->mce_mutex); - snd_wss_calibrate_mute(chip, 1); if (chip->hardware == WSS_HW_CS4231A || (chip->hardware & WSS_HW_CS4232_MASK)) { spin_lock_irqsave(&chip->reg_lock, flags); @@ -681,7 +669,6 @@ static void snd_wss_playback_format(struct snd_wss *chip, udelay(100); /* this seems to help */ snd_wss_mce_down(chip); } - snd_wss_calibrate_mute(chip, 0); mutex_unlock(&chip->mce_mutex); } @@ -693,7 +680,6 @@ static void snd_wss_capture_format(struct snd_wss *chip, int full_calib = 1; mutex_lock(&chip->mce_mutex); - snd_wss_calibrate_mute(chip, 1); if (chip->hardware == WSS_HW_CS4231A || (chip->hardware & WSS_HW_CS4232_MASK)) { spin_lock_irqsave(&chip->reg_lock, flags); @@ -750,7 +736,6 @@ static void snd_wss_capture_format(struct snd_wss *chip, spin_unlock_irqrestore(&chip->reg_lock, flags); snd_wss_mce_down(chip); } - snd_wss_calibrate_mute(chip, 0); mutex_unlock(&chip->mce_mutex); } @@ -807,6 +792,7 @@ static void snd_wss_init(struct snd_wss *chip) { unsigned long flags; + snd_wss_calibrate_mute(chip, 1); snd_wss_mce_down(chip); #ifdef SNDRV_DEBUG_MCE @@ -830,6 +816,8 @@ static void snd_wss_init(struct snd_wss *chip) snd_wss_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); + chip->image[CS4231_IFACE_CTRL] &= ~CS4231_AUTOCALIB; + snd_wss_out(chip, CS4231_IFACE_CTRL, chip->image[CS4231_IFACE_CTRL]); snd_wss_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1]); spin_unlock_irqrestore(&chip->reg_lock, flags); @@ -863,6 +851,7 @@ static void snd_wss_init(struct snd_wss *chip) chip->image[CS4231_REC_FORMAT]); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_wss_mce_down(chip); + snd_wss_calibrate_mute(chip, 0); #ifdef SNDRV_DEBUG_MCE snd_printk("init: (5)\n"); @@ -921,8 +910,6 @@ static void snd_wss_close(struct snd_wss *chip, unsigned int mode) mutex_unlock(&chip->open_mutex); return; } - snd_wss_calibrate_mute(chip, 1); - /* disable IRQ */ spin_lock_irqsave(&chip->reg_lock, flags); if (!(chip->hardware & WSS_HW_AD1848_MASK)) @@ -955,8 +942,6 @@ static void snd_wss_close(struct snd_wss *chip, unsigned int mode) wss_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_wss_calibrate_mute(chip, 0); - chip->mode = 0; mutex_unlock(&chip->open_mutex); } @@ -1149,7 +1134,7 @@ irqreturn_t snd_wss_interrupt(int irq, void *dev_id) if (chip->hardware & WSS_HW_AD1848_MASK) wss_outb(chip, CS4231P(STATUS), 0); else - snd_wss_outm(chip, CS4231_IRQ_STATUS, status, 0); + snd_wss_out(chip, CS4231_IRQ_STATUS, status); spin_unlock(&chip->reg_lock); return IRQ_HANDLED; } -- 1.5.2.2 ---------------------------------------------------------------------- Speak Up. Angielski szybko i skutecznie. 3 miesiace nauki gratis. Sprawdz. >> http://link.interia.pl/f2019

3 2

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 16 Jan '09

16 Jan '09

The following changes since commit c0106d72b8d71696dbe9dc80e2c77d4ac63f7531: Takashi Iwai (1): Merge branch 'topic/asoc' into next/asoc are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Ben Nizette (1): ASoC: atmel_pcm: Remove non-existant header Hugo Villeneuve (1): ASoC: DaVinci: Fix SFFSDR compilation error. Ian Molton (3): ASoC: Driver for the WM9705 AC97 codec. ASoC: machine driver for Toshiba e750 ASoC: machine driver for Toshiba e800 Mark Brown (1): Merge branch 'for-2.6.29' into for-2.6.30 arch/arm/mach-pxa/e750.c | 5 + arch/arm/mach-pxa/include/mach/eseries-gpio.h | 10 + sound/soc/atmel/atmel-pcm.c | 2 - sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 3 + sound/soc/codecs/wm9705.c | 410 +++++++++++++++++++++++++ sound/soc/codecs/wm9705.h | 14 + sound/soc/davinci/davinci-sffsdr.c | 20 +- sound/soc/pxa/Kconfig | 9 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/e750_wm9705.c | 189 ++++++++++++ sound/soc/pxa/e800_wm9712.c | 116 ++++++- 12 files changed, 765 insertions(+), 19 deletions(-) create mode 100644 sound/soc/codecs/wm9705.c create mode 100644 sound/soc/codecs/wm9705.h create mode 100644 sound/soc/pxa/e750_wm9705.c

2 5

[alsa-devel] ASoC update for 2.6.29
by Mark Brown 16 Jan '09

16 Jan '09

The following changes since commit eff317d0834ad1ff03f747f6bc2d76b9a9c95160: Peter Ujfalusi (1): ASoC: Fix the power update function for snd_soc_dapm_value_mux are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.29 Ben Nizette (1): ASoC: atmel_pcm: Remove non-existant header sound/soc/atmel/atmel-pcm.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-)

2 2

[alsa-devel] [PATCH 1/1] ASoC: Fix compilation error with davinci-sffsdr module
by Hugo Villeneuve 16 Jan '09

16 Jan '09

The SFFSDR audio codec sampling frequency clock is generated by a FPGA, and set by the FPGA driver. The FPGA driver is not yet in the mainline kernel, so that is why an extern declaration is used in the meantime. Signed-off-by: Hugo Villeneuve <hugo(a)hugovil.com> --- sound/soc/davinci/davinci-sffsdr.c | 6 +++++- 1 files changed, 5 insertions(+), 1 deletions(-) diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c index 1bbde3e..0a03b41 100644 --- a/sound/soc/davinci/davinci-sffsdr.c +++ b/sound/soc/davinci/davinci-sffsdr.c @@ -25,7 +25,6 @@ #include <asm/gpio.h> #include <asm/dma.h> #include <asm/mach-types.h> -#include <asm/plat-sffsdr/sffsdr-fpga.h> #include <mach/asp.h> #include <mach/edma.h> @@ -34,6 +33,11 @@ #include "davinci-pcm.h" #include "davinci-i2s.h" +/* The SFFSDR audio codec sampling frequency clock is generated by a FPGA. + * The FPGA driver is not yet in the mainline kernel, so that is + * why an extern declaration is used in the meantime. */ +extern int sffsdr_fpga_set_codec_fs(int fs); + static int sffsdr_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { -- 1.5.4.5

4 14

[alsa-devel] Creative X-Fi USB Driver Status
by Mandar Joshi 16 Jan '09

16 Jan '09

Hello, I would like to help developing a driver for the Creative X-Fi USB. Is someone working on it presently? I got the latest sources from the git repository. Didn't find any mention about the X-Fi USB in alsa-driver/usb/ . Regards Mandar Joshi

1 0

[alsa-devel] Creative X-Fi USB Driver Status
by Mandar Joshi 16 Jan '09

16 Jan '09

Hello, I would like to help developing a driver for the Creative X-Fi USB. Is someone working on it presently? I got the latest sources from the git repository. Didn't find any mention about the X-Fi USB in alsa-driver/usb/ . Regards Mandar Joshi

1 0

[alsa-devel] [PATCH] snd-usb-caiaq: support for two more audio devices
by Daniel Mack 16 Jan '09

16 Jan '09

- Added support for two new audio devices from Native Instuments, 'Audio4DJ' and 'GuitarRig mobile' - Add missing statement about 'Session IO' in Kconfig help text - Version number bumped to 1.3.11 Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- Resending an updated version - this one should apply cleanly to git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git sound/usb/Kconfig | 3 +++ sound/usb/caiaq/caiaq-audio.c | 5 +++-- sound/usb/caiaq/caiaq-control.c | 15 ++++++++++++--- sound/usb/caiaq/caiaq-device.c | 16 ++++++++++++++-- sound/usb/caiaq/caiaq-device.h | 2 ++ 5 files changed, 34 insertions(+), 7 deletions(-) diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 4f0eac9..523aec1 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -48,7 +48,10 @@ config SND_USB_CAIAQ * Native Instruments Kore Controller * Native Instruments Kore Controller 2 * Native Instruments Audio Kontrol 1 + * Native Instruments Audio 4 DJ * Native Instruments Audio 8 DJ + * Native Instruments Guitar Rig Session I/O + * Native Instruments Guitar Rig mobile To compile this driver as a module, choose M here: the module will be called snd-usb-caiaq. diff --git a/sound/usb/caiaq/caiaq-audio.c b/sound/usb/caiaq/caiaq-audio.c index b3a6033..fc6d571 100644 --- a/sound/usb/caiaq/caiaq-audio.c +++ b/sound/usb/caiaq/caiaq-audio.c @@ -638,9 +638,10 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AK1): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL3): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_SESSIONIO): - dev->samplerates |= SNDRV_PCM_RATE_88200; + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_GUITARRIGMOBILE): dev->samplerates |= SNDRV_PCM_RATE_192000; - break; + /* fall thru */ + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ): dev->samplerates |= SNDRV_PCM_RATE_88200; break; diff --git a/sound/usb/caiaq/caiaq-control.c b/sound/usb/caiaq/caiaq-control.c index ccd763d..6ac5489 100644 --- a/sound/usb/caiaq/caiaq-control.c +++ b/sound/usb/caiaq/caiaq-control.c @@ -39,14 +39,15 @@ static int control_info(struct snd_kcontrol *kcontrol, struct snd_usb_caiaqdev *dev = caiaqdev(chip->card); int pos = kcontrol->private_value; int is_intval = pos & CNT_INTVAL; + unsigned int id = dev->chip.usb_id; uinfo->count = 1; pos &= ~CNT_INTVAL; - if (dev->chip.usb_id == - USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ) + if (((id == USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ)) || + (id == USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ))) && (pos == 0)) { - /* current input mode of A8DJ */ + /* current input mode of A8DJ and A4DJ */ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->value.integer.min = 0; uinfo->value.integer.max = 2; @@ -247,6 +248,10 @@ static struct caiaq_controller a8dj_controller[] = { { "Software lock", 40 } }; +static struct caiaq_controller a4dj_controller[] = { + { "Current input mode", 0 | CNT_INTVAL } +}; + static int __devinit add_controls(struct caiaq_controller *c, int num, struct snd_usb_caiaqdev *dev) { @@ -295,6 +300,10 @@ int __devinit snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *dev) ret = add_controls(a8dj_controller, ARRAY_SIZE(a8dj_controller), dev); break; + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ): + ret = add_controls(a4dj_controller, + ARRAY_SIZE(a4dj_controller), dev); + break; } return ret; diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c index 09aed23..2464614 100644 --- a/sound/usb/caiaq/caiaq-device.c +++ b/sound/usb/caiaq/caiaq-device.c @@ -42,15 +42,17 @@ #endif MODULE_AUTHOR("Daniel Mack <daniel(a)caiaq.de>"); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.10"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.11"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," "{Native Instruments, Kore Controller}," "{Native Instruments, Kore Controller 2}," "{Native Instruments, Audio Kontrol 1}," + "{Native Instruments, Audio 4 DJ}," "{Native Instruments, Audio 8 DJ}," - "{Native Instruments, Session I/O}}"); + "{Native Instruments, Session I/O}," + "{Native Instruments, GuitarRig mobile}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */ @@ -116,6 +118,16 @@ static struct usb_device_id snd_usb_id_table[] = { .idVendor = USB_VID_NATIVEINSTRUMENTS, .idProduct = USB_PID_SESSIONIO }, + { + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = USB_VID_NATIVEINSTRUMENTS, + .idProduct = USB_PID_GUITARRIGMOBILE + }, + { + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = USB_VID_NATIVEINSTRUMENTS, + .idProduct = USB_PID_AUDIO4DJ + }, { /* terminator */ } }; diff --git a/sound/usb/caiaq/caiaq-device.h b/sound/usb/caiaq/caiaq-device.h index ab56e73..0560c32 100644 --- a/sound/usb/caiaq/caiaq-device.h +++ b/sound/usb/caiaq/caiaq-device.h @@ -10,8 +10,10 @@ #define USB_PID_KORECONTROLLER 0x4711 #define USB_PID_KORECONTROLLER2 0x4712 #define USB_PID_AK1 0x0815 +#define USB_PID_AUDIO4DJ 0x0839 #define USB_PID_AUDIO8DJ 0x1978 #define USB_PID_SESSIONIO 0x1915 +#define USB_PID_GUITARRIGMOBILE 0x0d8d #define EP1_BUFSIZE 64 #define CAIAQ_USB_STR_LEN 0xff -- 1.5.6.5

2 1

[alsa-devel] [PATCH] snd-usb-caiaq: support for two more audio devices
by Daniel Mack 16 Jan '09

16 Jan '09

- Added support for two new audio devices from Native Instuments, 'Audio4DJ' and 'GuitarRig mobile' - Version number bumped to 1.3.11 Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/Kconfig | 2 ++ sound/usb/caiaq/caiaq-audio.c | 5 +++-- sound/usb/caiaq/caiaq-control.c | 15 ++++++++++++--- sound/usb/caiaq/caiaq-device.c | 16 ++++++++++++++-- sound/usb/caiaq/caiaq-device.h | 2 ++ 5 files changed, 33 insertions(+), 7 deletions(-) diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 5905c2b..523aec1 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -48,8 +48,10 @@ config SND_USB_CAIAQ * Native Instruments Kore Controller * Native Instruments Kore Controller 2 * Native Instruments Audio Kontrol 1 + * Native Instruments Audio 4 DJ * Native Instruments Audio 8 DJ * Native Instruments Guitar Rig Session I/O + * Native Instruments Guitar Rig mobile To compile this driver as a module, choose M here: the module will be called snd-usb-caiaq. diff --git a/sound/usb/caiaq/caiaq-audio.c b/sound/usb/caiaq/caiaq-audio.c index b3a6033..fc6d571 100644 --- a/sound/usb/caiaq/caiaq-audio.c +++ b/sound/usb/caiaq/caiaq-audio.c @@ -638,9 +638,10 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AK1): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL3): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_SESSIONIO): - dev->samplerates |= SNDRV_PCM_RATE_88200; + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_GUITARRIGMOBILE): dev->samplerates |= SNDRV_PCM_RATE_192000; - break; + /* fall thru */ + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ): dev->samplerates |= SNDRV_PCM_RATE_88200; break; diff --git a/sound/usb/caiaq/caiaq-control.c b/sound/usb/caiaq/caiaq-control.c index ccd763d..6ac5489 100644 --- a/sound/usb/caiaq/caiaq-control.c +++ b/sound/usb/caiaq/caiaq-control.c @@ -39,14 +39,15 @@ static int control_info(struct snd_kcontrol *kcontrol, struct snd_usb_caiaqdev *dev = caiaqdev(chip->card); int pos = kcontrol->private_value; int is_intval = pos & CNT_INTVAL; + unsigned int id = dev->chip.usb_id; uinfo->count = 1; pos &= ~CNT_INTVAL; - if (dev->chip.usb_id == - USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ) + if (((id == USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ)) || + (id == USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ))) && (pos == 0)) { - /* current input mode of A8DJ */ + /* current input mode of A8DJ and A4DJ */ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->value.integer.min = 0; uinfo->value.integer.max = 2; @@ -247,6 +248,10 @@ static struct caiaq_controller a8dj_controller[] = { { "Software lock", 40 } }; +static struct caiaq_controller a4dj_controller[] = { + { "Current input mode", 0 | CNT_INTVAL } +}; + static int __devinit add_controls(struct caiaq_controller *c, int num, struct snd_usb_caiaqdev *dev) { @@ -295,6 +300,10 @@ int __devinit snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *dev) ret = add_controls(a8dj_controller, ARRAY_SIZE(a8dj_controller), dev); break; + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ): + ret = add_controls(a4dj_controller, + ARRAY_SIZE(a4dj_controller), dev); + break; } return ret; diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c index 41c36b0..d09fc2a 100644 --- a/sound/usb/caiaq/caiaq-device.c +++ b/sound/usb/caiaq/caiaq-device.c @@ -42,15 +42,17 @@ #endif MODULE_AUTHOR("Daniel Mack <daniel(a)caiaq.de>"); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.10"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.11"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," "{Native Instruments, Kore Controller}," "{Native Instruments, Kore Controller 2}," "{Native Instruments, Audio Kontrol 1}," + "{Native Instruments, Audio 4 DJ}," "{Native Instruments, Audio 8 DJ}," - "{Native Instruments, Session I/O}}"); + "{Native Instruments, Session I/O}," + "{Native Instruments, GuitarRig mobile}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */ @@ -116,6 +118,16 @@ static struct usb_device_id snd_usb_id_table[] = { .idVendor = USB_VID_NATIVEINSTRUMENTS, .idProduct = USB_PID_SESSIONIO }, + { + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = USB_VID_NATIVEINSTRUMENTS, + .idProduct = USB_PID_GUITARRIGMOBILE + }, + { + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = USB_VID_NATIVEINSTRUMENTS, + .idProduct = USB_PID_AUDIO4DJ + }, { /* terminator */ } }; diff --git a/sound/usb/caiaq/caiaq-device.h b/sound/usb/caiaq/caiaq-device.h index 9fb91af..7010651 100644 --- a/sound/usb/caiaq/caiaq-device.h +++ b/sound/usb/caiaq/caiaq-device.h @@ -10,8 +10,10 @@ #define USB_PID_KORECONTROLLER 0x4711 #define USB_PID_KORECONTROLLER2 0x4712 #define USB_PID_AK1 0x0815 +#define USB_PID_AUDIO4DJ 0x0839 #define USB_PID_AUDIO8DJ 0x1978 #define USB_PID_SESSIONIO 0x1915 +#define USB_PID_GUITARRIGMOBILE 0x0d8d #define EP1_BUFSIZE 64 #define CAIAQ_USB_STR_LEN 0xff -- 1.5.6.5

2 1

[alsa-devel] [PATCH] ALSA: hda - add quirks for some 82801H variants to use ALC883_MITAC
by Luke Yelavich 16 Jan '09

16 Jan '09

Add the 82801H variants 1071:8227 and 8086:2503 to use ALC883_MITAC Reference: Ubuntu bug 210865 https://bugs.launchpad.net/bugs/210865 Signed-off-by: Luke Yelavich <themuso(a)ubuntu.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 82dd084..5d249a5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8478,6 +8478,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_ASUS_EEE1601), SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG), SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1071, 0x8227, "Mitac 82801H", ALC883_MITAC), SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x10f1, 0x2350, "TYAN-S2350", ALC888_6ST_DELL), @@ -8526,6 +8527,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66), SND_PCI_QUIRK(0x8086, 0x0001, "DG33BUC", ALC883_3ST_6ch_INTEL), SND_PCI_QUIRK(0x8086, 0x0002, "DG33FBC", ALC883_3ST_6ch_INTEL), + SND_PCI_QUIRK(0x8086, 0x2503, "82801H", ALC883_MITAC), SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC883_3ST_6ch_INTEL), SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch), {} -- 1.6.0.4

2 1

[alsa-devel] [PATCH 1/1] ALSA: ASoC: DaVinci: Fix SFFSDR compilation error.
by Hugo Villeneuve 15 Jan '09

15 Jan '09

Remove dependency on sffsdr_fpga_set_codec_fs() when the SFFSDR FPGA module is not selected. Signed-off-by: Hugo Villeneuve <hugo(a)hugovil.com> --- sound/soc/davinci/davinci-sffsdr.c | 20 +++++++++++++++++--- 1 files changed, 17 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c index 3e8e6cd..e3ea07a 100644 --- a/sound/soc/davinci/davinci-sffsdr.c +++ b/sound/soc/davinci/davinci-sffsdr.c @@ -25,7 +25,9 @@ #include <asm/gpio.h> #include <asm/dma.h> #include <asm/mach-types.h> +#ifdef CONFIG_SFFSDR_FPGA #include <asm/plat-sffsdr/sffsdr-fpga.h> +#endif #include <mach/asp.h> #include <mach/edma.h> @@ -42,6 +44,17 @@ static int sffsdr_hw_params(struct snd_pcm_substream *substream, int fs; int ret = 0; + /* Fsref can be 32000, 44100 or 48000. */ + fs = params_rate(params); + +#ifndef CONFIG_SFFSDR_FPGA + /* Without the FPGA module, the Fs is fixed at 44100 Hz */ + if (fs != 44100) { + pr_debug("warning: only 44.1 kHz is supported without SFFSDR FPGA module\n"); + return -EINVAL; + } +#endif + /* Set cpu DAI configuration: * CLKX and CLKR are the inputs for the Sample Rate Generator. * FSX and FSR are outputs, driven by the sample Rate Generator. */ @@ -52,12 +65,13 @@ static int sffsdr_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - /* Fsref can be 32000, 44100 or 48000. */ - fs = params_rate(params); - pr_debug("sffsdr_hw_params: rate = %d Hz\n", fs); +#ifndef CONFIG_SFFSDR_FPGA + return 0; +#else return sffsdr_fpga_set_codec_fs(fs); +#endif } static struct snd_soc_ops sffsdr_ops = { -- 1.5.4.5

3 3

[alsa-devel] [PATCH 1/1] ALSA: ASoC: DaVinci: Fix SFFSDR init bug.
by Hugo Villeneuve 15 Jan '09

15 Jan '09

Removed call to machine_is_sffsdr() in module init function because it always returns zero and causes GCC to optimise and remove all functions of the module. Signed-off-by: Hugo Villeneuve <hugo(a)hugovil.com> --- sound/soc/davinci/davinci-sffsdr.c | 3 --- 1 files changed, 0 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c index 1bbde3e..3e8e6cd 100644 --- a/sound/soc/davinci/davinci-sffsdr.c +++ b/sound/soc/davinci/davinci-sffsdr.c @@ -115,9 +115,6 @@ static int __init sffsdr_init(void) { int ret; - if (!machine_is_sffsdr()) - return -EINVAL; - sffsdr_snd_device = platform_device_alloc("soc-audio", 0); if (!sffsdr_snd_device) { printk(KERN_ERR "platform device allocation failed\n"); -- 1.5.4.5

4 8

[alsa-devel] [PATCH] snd-aoa: handle older machines
by Johannes Berg 15 Jan '09

15 Jan '09

This patch changes snd-aoa to handle some older machines that are currently handled by snd-powermac. snd-aoa has a number of advantages though, notably it can autoload better and is generally a more modern driver. By hardcoding the accepted device-ids (last hunk of the patch) I'm trying to avoid regressions because this driver will otherwise load automatically and not let snd-powermac load. People who are unhappy with snd-powermac and have a device-id property in the device tree are encouraged to read this patch and make a patch to amend this as appropriate. Signed-off-by: Johannes Berg <johannes(a)sipsolutions.net> --- sound/aoa/fabrics/layout.c | 74 ++++++++++++++++++++++++++++++--------- sound/aoa/soundbus/i2sbus/core.c | 22 ++++++++--- 2 files changed, 74 insertions(+), 22 deletions(-) --- wireless-testing.orig/sound/aoa/fabrics/layout.c 2009-01-13 17:01:04.000000000 +0100 +++ wireless-testing/sound/aoa/fabrics/layout.c 2009-01-15 18:15:28.000000000 +0100 @@ -1,16 +1,14 @@ /* - * Apple Onboard Audio driver -- layout fabric + * Apple Onboard Audio driver -- layout/machine id fabric * - * Copyright 2006 Johannes Berg <johannes(a)sipsolutions.net> + * Copyright 2006-2008 Johannes Berg <johannes(a)sipsolutions.net> * * GPL v2, can be found in COPYING. * * - * This fabric module looks for sound codecs - * based on the layout-id property in the device tree. - * + * This fabric module looks for sound codecs based on the + * layout-id or device-id property in the device tree. */ - #include <asm/prom.h> #include <linux/list.h> #include <linux/module.h> @@ -63,7 +61,7 @@ struct codec_connect_info { #define LAYOUT_FLAG_COMBO_LINEOUT_SPDIF (1<<0) struct layout { - unsigned int layout_id; + unsigned int layout_id, device_id; struct codec_connect_info codecs[MAX_CODECS_PER_BUS]; int flags; @@ -111,6 +109,10 @@ MODULE_ALIAS("sound-layout-96"); MODULE_ALIAS("sound-layout-98"); MODULE_ALIAS("sound-layout-100"); +MODULE_ALIAS("aoa-device-id-14"); +MODULE_ALIAS("aoa-device-id-22"); +MODULE_ALIAS("aoa-device-id-35"); + /* onyx with all but microphone connected */ static struct codec_connection onyx_connections_nomic[] = { { @@ -518,6 +520,27 @@ static struct layout layouts[] = { .connections = onyx_connections_noheadphones, }, }, + /* PowerMac3,4 */ + { .device_id = 14, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_noline, + }, + }, + /* PowerMac3,6 */ + { .device_id = 22, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_all, + }, + }, + /* PowerBook5,2 */ + { .device_id = 35, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_all, + }, + }, {} }; @@ -526,7 +549,7 @@ static struct layout *find_layout_by_id( struct layout *l; l = layouts; - while (l->layout_id) { + while (l->codecs[0].name) { if (l->layout_id == id) return l; l++; @@ -534,6 +557,19 @@ static struct layout *find_layout_by_id( return NULL; } +static struct layout *find_layout_by_device(unsigned int id) +{ + struct layout *l; + + l = layouts; + while (l->codecs[0].name) { + if (l->device_id == id) + return l; + l++; + } + return NULL; +} + static void use_layout(struct layout *l) { int i; @@ -938,8 +974,8 @@ static struct aoa_fabric layout_fabric = static int aoa_fabric_layout_probe(struct soundbus_dev *sdev) { struct device_node *sound = NULL; - const unsigned int *layout_id; - struct layout *layout; + const unsigned int *id; + struct layout *layout = NULL; struct layout_dev *ldev = NULL; int err; @@ -952,15 +988,18 @@ static int aoa_fabric_layout_probe(struc if (sound->type && strcasecmp(sound->type, "soundchip") == 0) break; } - if (!sound) return -ENODEV; + if (!sound) + return -ENODEV; - layout_id = of_get_property(sound, "layout-id", NULL); - if (!layout_id) - goto outnodev; - printk(KERN_INFO "snd-aoa-fabric-layout: found bus with layout %d\n", - *layout_id); + id = of_get_property(sound, "layout-id", NULL); + if (id) { + layout = find_layout_by_id(*id); + } else { + id = of_get_property(sound, "device-id", NULL); + if (id) + layout = find_layout_by_device(*id); + } - layout = find_layout_by_id(*layout_id); if (!layout) { printk(KERN_ERR "snd-aoa-fabric-layout: unknown layout\n"); goto outnodev; @@ -976,6 +1015,7 @@ static int aoa_fabric_layout_probe(struc ldev->layout = layout; ldev->gpio.node = sound->parent; switch (layout->layout_id) { + case 0: /* anything with device_id, not layout_id */ case 41: /* that unknown machine no one seems to have */ case 51: /* PowerBook5,4 */ case 58: /* Mac Mini */ --- wireless-testing.orig/sound/aoa/soundbus/i2sbus/core.c 2009-01-13 17:01:04.000000000 +0100 +++ wireless-testing/sound/aoa/soundbus/i2sbus/core.c 2009-01-15 18:15:34.000000000 +0100 @@ -1,7 +1,7 @@ /* * i2sbus driver * - * Copyright 2006 Johannes Berg <johannes(a)sipsolutions.net> + * Copyright 2006-2008 Johannes Berg <johannes(a)sipsolutions.net> * * GPL v2, can be found in COPYING. */ @@ -186,13 +186,25 @@ static int i2sbus_add_dev(struct macio_d } } if (i == 1) { - const u32 *layout_id = - of_get_property(sound, "layout-id", NULL); - if (layout_id) { - layout = *layout_id; + const u32 *id = of_get_property(sound, "layout-id", NULL); + + if (id) { + layout = *id; snprintf(dev->sound.modalias, 32, "sound-layout-%d", layout); ok = 1; + } else { + id = of_get_property(sound, "device-id", NULL); + /* + * We probably cannot handle all device-id machines, + * so restrict to those we do handle for now. + */ + if (id && (*id == 22 || *id == 14 || *id == 35)) { + snprintf(dev->sound.modalias, 32, + "aoa-device-id-%d", *id); + ok = 1; + layout = -1; + } } } /* for the time being, until we can handle non-layout-id

2 1

[alsa-devel] [PATCH] snd-aoa: handle master-amp if present
by Johannes Berg 15 Jan '09

15 Jan '09

Some machines have a master amp GPIO that needs to be toggled to get sound output, in addition to speaker/headphone/line-out amps. This makes snd-aoa handle it, if present in the device tree, thus making snd-aoa be able to output sound on PowerMac3,6, which was previously handled by snd-powermac which also doesn't use the master amp GPIO. Signed-off-by: Johannes Berg <johannes(a)sipsolutions.net> --- sound/aoa/aoa-gpio.h | 2 ++ sound/aoa/core/gpio-feature.c | 17 ++++++++++++++++- sound/aoa/fabrics/layout.c | 7 +++++++ 3 files changed, 25 insertions(+), 1 deletion(-) --- wireless-testing.orig/sound/aoa/aoa-gpio.h 2009-01-15 18:09:48.000000000 +0100 +++ wireless-testing/sound/aoa/aoa-gpio.h 2009-01-15 18:15:47.000000000 +0100 @@ -34,10 +34,12 @@ struct gpio_methods { void (*set_headphone)(struct gpio_runtime *rt, int on); void (*set_speakers)(struct gpio_runtime *rt, int on); void (*set_lineout)(struct gpio_runtime *rt, int on); + void (*set_master)(struct gpio_runtime *rt, int on); int (*get_headphone)(struct gpio_runtime *rt); int (*get_speakers)(struct gpio_runtime *rt); int (*get_lineout)(struct gpio_runtime *rt); + int (*get_master)(struct gpio_runtime *rt); void (*set_hw_reset)(struct gpio_runtime *rt, int on); --- wireless-testing.orig/sound/aoa/core/gpio-feature.c 2009-01-15 18:09:48.000000000 +0100 +++ wireless-testing/sound/aoa/core/gpio-feature.c 2009-01-15 18:18:09.000000000 +0100 @@ -14,7 +14,7 @@ #include <linux/interrupt.h> #include "../aoa.h" -/* TODO: these are 20 global variables +/* TODO: these are lots of global variables * that aren't used on most machines... * Move them into a dynamically allocated * structure and use that. @@ -23,6 +23,7 @@ /* these are the GPIO numbers (register addresses as offsets into * the GPIO space) */ static int headphone_mute_gpio; +static int master_mute_gpio; static int amp_mute_gpio; static int lineout_mute_gpio; static int hw_reset_gpio; @@ -32,6 +33,7 @@ static int linein_detect_gpio; /* see the SWITCH_GPIO macro */ static int headphone_mute_gpio_activestate; +static int master_mute_gpio_activestate; static int amp_mute_gpio_activestate; static int lineout_mute_gpio_activestate; static int hw_reset_gpio_activestate; @@ -156,6 +158,7 @@ static int ftr_gpio_get_##name(struct gp FTR_GPIO(headphone, 0); FTR_GPIO(amp, 1); FTR_GPIO(lineout, 2); +FTR_GPIO(master, 3); static void ftr_gpio_set_hw_reset(struct gpio_runtime *rt, int on) { @@ -172,6 +175,8 @@ static void ftr_gpio_set_hw_reset(struct hw_reset_gpio, v); } +static struct gpio_methods methods; + static void ftr_gpio_all_amps_off(struct gpio_runtime *rt) { int saved; @@ -181,6 +186,8 @@ static void ftr_gpio_all_amps_off(struct ftr_gpio_set_headphone(rt, 0); ftr_gpio_set_amp(rt, 0); ftr_gpio_set_lineout(rt, 0); + if (methods.set_master) + ftr_gpio_set_master(rt, 0); rt->implementation_private = saved; } @@ -193,6 +200,8 @@ static void ftr_gpio_all_amps_restore(st ftr_gpio_set_headphone(rt, (s>>0)&1); ftr_gpio_set_amp(rt, (s>>1)&1); ftr_gpio_set_lineout(rt, (s>>2)&1); + if (methods.set_master) + ftr_gpio_set_master(rt, (s>>3)&1); } static void ftr_handle_notify(struct work_struct *work) @@ -231,6 +240,12 @@ static void ftr_gpio_init(struct gpio_ru get_gpio("hw-reset", "audio-hw-reset", &hw_reset_gpio, &hw_reset_gpio_activestate); + if (get_gpio("master-mute", NULL, + &master_mute_gpio, + &master_mute_gpio_activestate)) { + methods.set_master = ftr_gpio_set_master; + methods.get_master = ftr_gpio_get_master; + } headphone_detect_node = get_gpio("headphone-detect", NULL, &headphone_detect_gpio, --- wireless-testing.orig/sound/aoa/fabrics/layout.c 2009-01-15 18:15:28.000000000 +0100 +++ wireless-testing/sound/aoa/fabrics/layout.c 2009-01-15 18:15:47.000000000 +0100 @@ -600,6 +600,7 @@ struct layout_dev { struct snd_kcontrol *headphone_ctrl; struct snd_kcontrol *lineout_ctrl; struct snd_kcontrol *speaker_ctrl; + struct snd_kcontrol *master_ctrl; struct snd_kcontrol *headphone_detected_ctrl; struct snd_kcontrol *lineout_detected_ctrl; @@ -651,6 +652,7 @@ static struct snd_kcontrol_new n##_ctl = AMP_CONTROL(headphone, "Headphone Switch"); AMP_CONTROL(speakers, "Speakers Switch"); AMP_CONTROL(lineout, "Line-Out Switch"); +AMP_CONTROL(master, "Master Switch"); static int detect_choice_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -891,6 +893,11 @@ static void layout_attached_codec(struct lineout = codec->gpio->methods->get_detect(codec->gpio, AOA_NOTIFY_LINE_OUT); + if (codec->gpio->methods->set_master) { + ctl = snd_ctl_new1(&master_ctl, codec->gpio); + ldev->master_ctrl = ctl; + aoa_snd_ctl_add(ctl); + } while (cc->connected) { if (cc->connected & CC_SPEAKERS) { if (headphones <= 0 && lineout <= 0)

1 0

[alsa-devel] Mac Pro problem with USB soundcard
by James Ward 15 Jan '09

15 Jan '09

I'm running 64 Bit Sabayon Linux 4.0 r1 on a Mac Pro. I have connected a Creative USB sound card and would like to use it as the default. I tried these two .asoundrc files suggested by gnubien on #alsa: pcm.!default { type hw card 1 } ctl.!default { type hw card 1 } and: created by wishie on #alsa 10-13-07 #Set the card you want to use here.. this example is for hw:1,0 pcm.snd_card { type hw card 1 #change "card 1" to the card number listed for your usb soundcard using cat /proc/asound/cards output device 0 } pcm.!default { type plug slave.pcm "dmixer" } pcm.dmixer { type dmix ipc_key 1024 ipc_key_add_uid false ipc_perm 0660 slave { pcm "snd_card" rate 48000 period_time 0 period_size 1024 buffer_time 0 buffer_size 5120 } } Either of these .asoundrc files result in the USB soundcard going away. I have to unplug it and plug it back in to get it back. Please help. James -- Ward... James Ward Linux System Administrator System Administration Phone: 520-290-0910 ext 268 Fax: 520-546-3442 ICQ: 201663408 Jabber: james(a)jabber.chinoc.net

1 0

[alsa-devel] RFC: snd_card_create() function
by Takashi Iwai 15 Jan '09

15 Jan '09

Hi, so far, we use snd_card_new() function to create a card instance. A known problem regarding this API is that it doesn't return a proper error code, thus the probe callback always returns -ENOMEM (or whatever the driver defines) no matter which error occurred actually. A typical case is the card slot conflict. Even in such a case, it returns -ENOMEM, which may mislead the user. For fixing this situation, I'd like to convert from snd_card_new() to a new function, snd_card_create(): int snd_card_create(int idx, const char *id, struct module *module, int extra_size, struct snd_card **card_ret); and provides snd_card_new() as a wrapper with deprecated flag: static inline __deprecated struct snd_card *snd_card_new(int idx, const char *id, struct module *module, int extra_size) { struct snd_card *card; if (snd_card_create(idx, id, module, extra_size, &card) < 0) return NULL; return card; } The merit of creating a new function is that the older and out-of-tree codes (like xfi) work without modification. There is alternative option, namely to use PTR_ERR() and keep the function as is. But this has no advantage in practice -- it can't detect the old API at build time, and it may Oops. The preliminary patches are found in test/snd_card_new-err branch of sound git tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git test/snd_card_new-err If anyone finds problems with this, please let me know. thanks, Takashi

4 9

[alsa-devel] usb-audio extension unit
by James Courtier-Dutton 15 Jan '09

15 Jan '09

Hi, Is there any chance that someone could add get/set functions for usb audio "extension units". This would make adding support for usb devices such as the emu usb products a lot easier to do. Kind Regards James

2 1

[alsa-devel] [PATCH 0/1] ASoC: fix the dapm_mux_update_power function for value_mux
by Peter Ujfalusi 15 Jan '09

15 Jan '09

Hello, To be honest I don't know how I have missed this from the previous series with the cleanup for the value_mux type. the dapm_mux_update_power function needs to check for both snd_soc_dapm_mux and snd_soc_dapm_value_mux. --- Peter Ujfalusi (1): ASoC: Fix the power update function for snd_soc_dapm_value_mux sound/soc/soc-dapm.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-)

2 2

Re: [alsa-devel] [Fwd: Re: Fwd: audio ovolume problem on powerbook G3 Lombard (PPC740)]
by Risto Suominen 15 Jan '09

15 Jan '09

2009/1/8 Takashi Iwai <tiwai(a)suse.de>: > At Thu, 8 Jan 2009 14:06:05 +0200, > Risto Suominen wrote: >> >> 2009/1/8, Takashi Iwai <tiwai(a)suse.de>: >> > >> > We may use a virtual master control when all slave controls have the >> > identical volume resolution. >> > >> >> Sounds good. Is it something configurable, so there is no need to >> change the code? > > Look at sound/core/vmaster.c. > > > Takashi > Ok, I've tried to implement this on snd-powermac. Takashi, looks like there is a side effect: plugging in headphone doesn't show on alsamixer, even though it works, i.e. the speaker is silenced. Or have I forgotten something in my implementation? http://ristosu.wippiespace.com/pub/alsa-awacs-1.0.18a-p13.diff Richard, maybe you can try this? You should adjust Speaker and Line out to 100% in order to have full scale in Master. Risto

2 3

[alsa-devel] lockdep warning on 2.6.29-rc1 with pulseaudio
by Johannes Berg 15 Jan '09

15 Jan '09

Sorry if you've seen this already, I couldn't find it on gmane. soemthing is doing mutex_lock(&pcm->open_mutex); but this lock depends on &chip->open_mutex which in turn depends on &codec->spdif_mutex (seems a little odd) and that already depends on mmap_sem, through might_fault()? [ 49.742444] ======================================================= [ 49.742449] [ INFO: possible circular locking dependency detected ] [ 49.742453] 2.6.29-rc1-wl-11361-g269245b-dirty #15 [ 49.742455] ------------------------------------------------------- [ 49.742458] pulseaudio/3550 is trying to acquire lock: [ 49.742461] (&pcm->open_mutex){--..}, at: [<ffffffffa059f86e>] snd_pcm_release+0x3e/0xb0 [snd_pcm] [ 49.742479] [ 49.742479] but task is already holding lock: [ 49.742482] (&mm->mmap_sem){----}, at: [<ffffffff802c4a42>] sys_munmap+0x42/0x80 [ 49.742493] [ 49.742493] which lock already depends on the new lock. [ 49.742494] [ 49.742497] [ 49.742497] the existing dependency chain (in reverse order) is: [ 49.742500] [ 49.742501] -> #3 (&mm->mmap_sem){----}: [ 49.742507] [<ffffffff80273427>] check_prev_add+0x3b7/0x770 [ 49.742513] [<ffffffff80273de6>] validate_chain+0x606/0x6c0 [ 49.742518] [<ffffffff802742df>] __lock_acquire+0x43f/0xa10 [ 49.742522] [<ffffffff80274941>] lock_acquire+0x91/0xc0 [ 49.742526] [<ffffffff802bc39f>] might_fault+0x7f/0xb0 [ 49.742532] [<ffffffffa07fb15d>] snd_hda_mixer_amp_tlv+0x6d/0xf0 [snd_hda_codec] [ 49.742544] [<ffffffffa0b20bb0>] alc_cap_vol_tlv+0x70/0xa0 [snd_hda_codec_realtek] [ 49.742557] [<ffffffffa03a30be>] snd_ctl_tlv_ioctl+0x14e/0x1d0 [snd] [ 49.742571] [<ffffffffa03a527b>] snd_ctl_ioctl+0x26b/0x540 [snd] [ 49.742581] [<ffffffff802e82d1>] vfs_ioctl+0x31/0xa0 [ 49.742587] [<ffffffff802e8699>] do_vfs_ioctl+0x89/0x350 [ 49.742592] [<ffffffff802e89aa>] sys_ioctl+0x4a/0x80 [ 49.742597] [<ffffffff8020bd1b>] system_call_fastpath+0x16/0x1b [ 49.742603] [<ffffffffffffffff>] 0xffffffffffffffff [ 49.742608] [ 49.742609] -> #2 (&codec->spdif_mutex){--..}: [ 49.742615] [<ffffffff80273427>] check_prev_add+0x3b7/0x770 [ 49.742619] [<ffffffff80273de6>] validate_chain+0x606/0x6c0 [ 49.742623] [<ffffffff802742df>] __lock_acquire+0x43f/0xa10 [ 49.742628] [<ffffffff80274941>] lock_acquire+0x91/0xc0 [ 49.742632] [<ffffffff805ac5fc>] mutex_lock_nested+0xec/0x380 [ 49.742639] [<ffffffffa07f8d54>] snd_hda_multi_out_analog_open+0x84/0x130 [snd_hda_codec] [ 49.742649] [<ffffffffa0b1ac60>] alc880_playback_pcm_open+0x20/0x30 [snd_hda_codec_realtek] [ 49.742659] [<ffffffffa0897f39>] azx_pcm_open+0x209/0x280 [snd_hda_intel] [ 49.742667] [<ffffffffa05a01b8>] snd_pcm_open_substream+0x58/0xa0 [snd_pcm] [ 49.742677] [<ffffffffa05a038b>] snd_pcm_open+0x18b/0x280 [snd_pcm] [ 49.742687] [<ffffffffa05a04ec>] snd_pcm_playback_open+0x2c/0x40 [snd_pcm] [ 49.742697] [<ffffffffa039f7fc>] __snd_open+0x14c/0x270 [snd] [ 49.742707] [<ffffffffa039f947>] snd_open+0x27/0x40 [snd] [ 49.742717] [<ffffffff802dcd47>] chrdev_open+0x137/0x210 [ 49.742722] [<ffffffff802d7a5a>] __dentry_open+0x10a/0x340 [ 49.742727] [<ffffffff802d7d97>] nameidata_to_filp+0x57/0x70 [ 49.742732] [<ffffffff802e69de>] do_filp_open+0x1fe/0x970 [ 49.742737] [<ffffffff802d7868>] do_sys_open+0x78/0x110 [ 49.742742] [<ffffffff802d792b>] sys_open+0x1b/0x20 [ 49.742746] [<ffffffff8020bd1b>] system_call_fastpath+0x16/0x1b [ 49.742751] [<ffffffffffffffff>] 0xffffffffffffffff [ 49.742756] [ 49.742756] -> #1 (&chip->open_mutex){--..}: [ 49.742762] [<ffffffff80273427>] check_prev_add+0x3b7/0x770 [ 49.742767] [<ffffffff80273de6>] validate_chain+0x606/0x6c0 [ 49.742771] [<ffffffff802742df>] __lock_acquire+0x43f/0xa10 [ 49.742775] [<ffffffff80274941>] lock_acquire+0x91/0xc0 [ 49.742780] [<ffffffff805ac5fc>] mutex_lock_nested+0xec/0x380 [ 49.742785] [<ffffffffa0897d78>] azx_pcm_open+0x48/0x280 [snd_hda_intel] [ 49.742792] [<ffffffffa05a01b8>] snd_pcm_open_substream+0x58/0xa0 [snd_pcm] [ 49.742802] [<ffffffffa05a038b>] snd_pcm_open+0x18b/0x280 [snd_pcm] [ 49.742812] [<ffffffffa05a04ec>] snd_pcm_playback_open+0x2c/0x40 [snd_pcm] [ 49.742822] [<ffffffffa039f7fc>] __snd_open+0x14c/0x270 [snd] [ 49.742832] [<ffffffffa039f947>] snd_open+0x27/0x40 [snd] [ 49.742841] [<ffffffff802dcd47>] chrdev_open+0x137/0x210 [ 49.742846] [<ffffffff802d7a5a>] __dentry_open+0x10a/0x340 [ 49.742850] [<ffffffff802d7d97>] nameidata_to_filp+0x57/0x70 [ 49.742855] [<ffffffff802e69de>] do_filp_open+0x1fe/0x970 [ 49.742860] [<ffffffff802d7868>] do_sys_open+0x78/0x110 [ 49.742864] [<ffffffff802d792b>] sys_open+0x1b/0x20 [ 49.742869] [<ffffffff8020bd1b>] system_call_fastpath+0x16/0x1b [ 49.742874] [<ffffffffffffffff>] 0xffffffffffffffff [ 49.742882] [ 49.742883] -> #0 (&pcm->open_mutex){--..}: [ 49.742889] [<ffffffff802730c7>] check_prev_add+0x57/0x770 [ 49.742893] [<ffffffff80273de6>] validate_chain+0x606/0x6c0 [ 49.742897] [<ffffffff802742df>] __lock_acquire+0x43f/0xa10 [ 49.742902] [<ffffffff80274941>] lock_acquire+0x91/0xc0 [ 49.742906] [<ffffffff805ac5fc>] mutex_lock_nested+0xec/0x380 [ 49.742911] [<ffffffffa059f86e>] snd_pcm_release+0x3e/0xb0 [snd_pcm] [ 49.742921] [<ffffffff802dae20>] __fput+0xd0/0x1e0 [ 49.742926] [<ffffffff802daf4d>] fput+0x1d/0x30 [ 49.742931] [<ffffffff802c377a>] remove_vma+0x4a/0x80 [ 49.742936] [<ffffffff802c49b2>] do_munmap+0x342/0x390 [ 49.742940] [<ffffffff802c4a50>] sys_munmap+0x50/0x80 [ 49.742944] [<ffffffff8020bd1b>] system_call_fastpath+0x16/0x1b [ 49.742949] [<ffffffffffffffff>] 0xffffffffffffffff [ 49.742954] [ 49.742954] other info that might help us debug this: [ 49.742955] [ 49.742958] 1 lock held by pulseaudio/3550: [ 49.742961] #0: (&mm->mmap_sem){----}, at: [<ffffffff802c4a42>] sys_munmap+0x42/0x80 [ 49.742969] [ 49.742970] stack backtrace: [ 49.742973] Pid: 3550, comm: pulseaudio Not tainted 2.6.29-rc1-wl-11361-g269245b-dirty #15 [ 49.742976] Call Trace: [ 49.742981] [<ffffffff80272b70>] print_circular_bug_tail+0xe0/0xf0 [ 49.742985] [<ffffffff802730c7>] check_prev_add+0x57/0x770 [ 49.742991] [<ffffffff8026f414>] ? get_lock_stats+0x34/0x70 [ 49.742995] [<ffffffff80273de6>] validate_chain+0x606/0x6c0 [ 49.742998] [<ffffffff802742df>] __lock_acquire+0x43f/0xa10 [ 49.743002] [<ffffffff80274941>] lock_acquire+0x91/0xc0 [ 49.743012] [<ffffffffa059f86e>] ? snd_pcm_release+0x3e/0xb0 [snd_pcm] [ 49.743017] [<ffffffff805ac5fc>] mutex_lock_nested+0xec/0x380 [ 49.743027] [<ffffffffa059f86e>] ? snd_pcm_release+0x3e/0xb0 [snd_pcm] [ 49.743036] [<ffffffffa059f86e>] ? snd_pcm_release+0x3e/0xb0 [snd_pcm] [ 49.743046] [<ffffffffa059f86e>] snd_pcm_release+0x3e/0xb0 [snd_pcm] [ 49.743051] [<ffffffff802dae20>] __fput+0xd0/0x1e0 [ 49.743055] [<ffffffff802daf4d>] fput+0x1d/0x30 [ 49.743059] [<ffffffff802c377a>] remove_vma+0x4a/0x80 [ 49.743063] [<ffffffff802c49b2>] do_munmap+0x342/0x390 [ 49.743067] [<ffffffff802c4a50>] sys_munmap+0x50/0x80 [ 49.743071] [<ffffffff8020bd1b>] system_call_fastpath+0x16/0x1b Incidentally, sound doesn't work on my macbook5,1, I get lots of controls but no output. johannes

2 13

[alsa-devel] [PATCH] hda: fixed invalid value for SND_PCI_QUIRK
by Matthew Ranostay 14 Jan '09

14 Jan '09

Changed invalid STAC_92HD71BXX_REF value for reference board SND_PCI_QUIRK to correct STAC_92HD83XXX_REF. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index 35b83dc..a4e2d8f 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -1682,7 +1682,7 @@ static const char *stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, - "DFI LanParty", STAC_92HD71BXX_REF), + "DFI LanParty", STAC_92HD83XXX_REF), {} /* terminator */ };

2 1

[alsa-devel] Creative X-Fi not recognized since 1/9 build
by Susan Cragin 14 Jan '09

14 Jan '09

This is FYI. I have a creative X-fi soundcard and have been compiling developer builds for a couple of weeks, mostly with great success. But as of the 1/9 build, my X-fi card is no longer recognized. The driver it uses is CA0110. I do not re-compile kernel. Susan FYI PS -- About sound quality... I had been running Dragon NaturallySpeaking 10 on Linux through wine using the driver. The program measures accuracy for dictated sounds, and my scores have been pretty good, running from 28-31. With the hda-intel they are 25-28.

2 1