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[PATCH] Added double ac97 reset in pxa27x resume path.
by Robert Jarzmik 28 Apr '08

28 Apr '08

Signed-off-by: Robert Jarzmik <rjarzmik(a)free.fr> --- sound/soc/codecs/wm9713.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 8ec7ce9..5defac8 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -977,6 +977,8 @@ static int wm9713_resume(struct platform_device *pdev) /* give the codec an AC97 warm reset to start the link */ codec->ac97->bus->ops->reset(codec->ac97); codec->ac97->bus->ops->warm_reset(codec->ac97); + codec->ac97->bus->ops->reset(codec->ac97); + codec->ac97->bus->ops->warm_reset(codec->ac97); id = codec->soc_phys_read(codec->ac97, AC97_VENDOR_ID2); if (id != 0x4c13) { printk(KERN_ERR "wm9713 failed to resume"); -- 1.5.5.3 --=-=-= Content-Type: text/x-diff Content-Disposition: inline; filename=0004-Include-shuffle-in-2.6.25-kernels.patch

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[alsa-devel] Model for Toshiba Satellite L30 with Realtek ALC861-VD
by Roman Rybalko 28 Apr '08

28 Apr '08

I have the Toshiba L30 laptop with ALC861-VD codec. Auto detection doesn't enables integrated speakers (and headphones are detected as line-out). The suitable models for it is "dallas" and "linovo", but they create unusable widgets and don't provide full functionality. So I decided to write own model. I've token some usable parts from dallas and linovo and place they in my model, "toshiba". It's in alsa-driver-1.0.16.diff But I'd like to mute my microphone when it's unplugged (preferably by mixer), and also mute my speaker only by mixer when the HPs plugged in (now it is being muted by just disabling it's EAPD, as far as i understand), so I will be able to enable it any time even when HP is plugged. I've tried to do the first task - alsa-driver-1.0.16-micmute.diff, but I failed to record anything from microphone then. Could anyone give me an advice how should I do this? -- WBR, Roman Rybalko Codec: Realtek ALC861-VD Address: 0 Vendor Id: 0x10ec0862 Subsystem Id: 0x1179820d Revision Id: 0x100001 No Modem Function Group found Default PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x3c 0x40] PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x00 0x00] PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x00 0x00] PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital PCM: rates [0x160]: 44100 48000 96000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x09, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x1f 0x1f] PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x99 0x99] [0x1f 0x1f] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x01 0x01] Connection: 6 0x18 0x19 0x1a 0x1b 0x1c 0x1d Node 0x0c [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Connection: 2 0x04 0x0b Node 0x0f [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Connection: 2 0x05 0x0b Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x081003c: IN OUT HP EAPD Detect Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x0c Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0810034: IN OUT EAPD Detect Pin Default 0x01014020: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green Pin-ctls: 0x40: OUT Connection: 1 0x0d Node 0x16 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0834: IN OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x40: OUT Connection: 1 0x0e Node 0x17 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0834: IN OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x40: OUT Connection: 1 0x0f Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x081734: IN OUT Detect Pin Default 0x01a19830: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink Pin-ctls: 0x21: IN Connection: 1 0x0e Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08173c: IN OUT HP Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x21: IN Connection: 2 0x0c* 0x0f Node 0x1a [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0834: IN OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x20: IN Connection: 1 0x0d Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08173c: IN OUT HP Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x20: IN Connection: 2 0x0c* 0x0f Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x4014820d: [N/A] Speaker at Ext N/A Conn = RCA, Color = Purple Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x0810: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x23 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x25 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x26 [Vendor Defined Widget] wcaps 0xf00000: Mono 00:00.0 Host bridge [0600]: ATI Technologies Inc Unknown device [1002:5a31] (rev 01) Subsystem: Toshiba America Info Systems Unknown device [1179:ff31] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap- 66MHz+ UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort+ >SERR- <PERR- Latency: 64 00:01.0 PCI bridge [0604]: ATI Technologies Inc RS480 PCI Bridge [1002:5a3f] (prog-if 00 [Normal decode]) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz+ UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 64 Bus: primary=00, secondary=01, subordinate=01, sec-latency=64 I/O behind bridge: 00009000-00009fff Memory behind bridge: c0000000-c00fffff Prefetchable memory behind bridge: d0000000-dfffffff Secondary status: 66MHz+ FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort+ <SERR- <PERR- BridgeCtl: Parity- SERR- NoISA+ VGA+ MAbort- >Reset- FastB2B- Capabilities: <access denied> 00:12.0 IDE interface [0101]: ATI Technologies Inc 4379 Serial ATA Controller [1002:4379] (rev 80) (prog-if 8f [Master SecP SecO PriP PriO]) Subsystem: Toshiba America Info Systems Unknown device [1179:ff31] Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 64, Cache Line Size: 32 bytes Interrupt: pin A routed to IRQ 18 Region 0: I/O ports at 8440 [size=8] Region 1: I/O ports at 8430 [size=4] Region 2: I/O ports at 8420 [size=8] Region 3: I/O ports at 8410 [size=4] Region 4: I/O ports at 8400 [size=16] Region 5: Memory at c0407000 (32-bit, non-prefetchable) [size=512] [virtual] Expansion ROM at 44000000 [disabled] [size=512K] Capabilities: <access denied> 00:13.0 USB Controller [0c03]: ATI Technologies Inc IXP SB400 USB Host Controller [1002:4374] (rev 80) (prog-if 10 [OHCI]) Subsystem: Toshiba America Info Systems Unknown device [1179:ff31] Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV+ VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap- 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 64, Cache Line Size: 32 bytes Interrupt: pin A routed to IRQ 17 Region 0: Memory at c0404000 (32-bit, non-prefetchable) [size=4K] 00:13.1 USB Controller [0c03]: ATI Technologies Inc IXP SB400 USB Host Controller [1002:4375] (rev 80) (prog-if 10 [OHCI]) Subsystem: Toshiba America Info Systems Unknown device [1179:ff31] Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV+ VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap- 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 64, Cache Line Size: 32 bytes Interrupt: pin A routed to IRQ 17 Region 0: Memory at c0405000 (32-bit, non-prefetchable) [size=4K] 00:13.2 USB Controller [0c03]: ATI Technologies Inc IXP SB400 USB2 Host Controller [1002:4373] (rev 80) (prog-if 20 [EHCI]) Subsystem: Toshiba America Info Systems Unknown device [1179:ff31] Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV+ VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 64, Cache Line Size: 32 bytes Interrupt: pin A routed to IRQ 17 Region 0: Memory at c0406000 (32-bit, non-prefetchable) [size=4K] Capabilities: <access denied> 00:14.0 SMBus [0c05]: ATI Technologies Inc IXP SB400 SMBus Controller [1002:4372] (rev 82) Subsystem: Toshiba America Info Systems Unknown device [1179:ff31] Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap- 66MHz+ UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- Region 0: I/O ports at 8040 [size=16] Region 1: Memory at c0407400 (32-bit, non-prefetchable) [size=1K] 00:14.1 IDE interface [0101]: ATI Technologies Inc Standard Dual Channel PCI IDE Controller [1002:4376] (rev 80) (prog-if 88 [Master SecP]) Subsystem: Toshiba America Info Systems Unknown device [1179:ff31] Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV+ VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz+ UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 0 Interrupt: pin A routed to IRQ 19 Region 0: I/O ports at 01f0 [size=8] Region 1: I/O ports at 03f4 [size=1] Region 2: I/O ports at 0170 [size=8] Region 3: I/O ports at 0374 [size=1] Region 4: I/O ports at 8460 [size=16] Capabilities: <access denied> 00:14.2 Audio device [0403]: ATI Technologies Inc SB450 HDA Audio [1002:437b] (rev 01) Subsystem: Toshiba America Info Systems Unknown device [1179:ff31] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=slow >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 64, Cache Line Size: 32 bytes Interrupt: pin ? routed to IRQ 19 Region 0: Memory at c0400000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> 00:14.3 ISA bridge [0601]: ATI Technologies Inc IXP SB400 PCI-ISA Bridge [1002:4377] (rev 80) Subsystem: Toshiba America Info Systems Unknown device [1179:ff31] Control: I/O+ Mem+ BusMaster+ SpecCycle+ MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap- 66MHz+ UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 0 00:14.4 PCI bridge [0604]: ATI Technologies Inc IXP SB400 PCI-PCI Bridge [1002:4371] (rev 80) (prog-if 01 [Subtractive decode]) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap- 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 64 Bus: primary=00, secondary=09, subordinate=0e, sec-latency=64 I/O behind bridge: 0000a000-0000afff Memory behind bridge: c0100000-c01fffff Prefetchable memory behind bridge: 40000000-43ffffff Secondary status: 66MHz- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort+ <SERR- <PERR- BridgeCtl: Parity- SERR+ NoISA- VGA- MAbort- >Reset- FastB2B- 01:05.0 VGA compatible controller [0300]: ATI Technologies Inc RC410 [Radeon Xpress 200M] [1002:5a62] (prog-if 00 [VGA]) Subsystem: Toshiba America Info Systems Unknown device [1179:ff31] Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 255 (2000ns min), Cache Line Size: 32 bytes Interrupt: pin A routed to IRQ 20 Region 0: Memory at d0000000 (32-bit, prefetchable) [size=256M] Region 1: I/O ports at 9000 [size=256] Region 2: Memory at c0000000 (32-bit, non-prefetchable) [size=64K] [virtual] Expansion ROM at c0020000 [disabled] [size=128K] Capabilities: <access denied> 09:01.0 CardBus bridge [0607]: ENE Technology Inc CB1410 Cardbus Controller [1524:1410] (rev 01) Subsystem: Toshiba America Info Systems Unknown device [1179:ff31] Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 168, Cache Line Size: 128 bytes Interrupt: pin A routed to IRQ 16 Region 0: Memory at c0101000 (32-bit, non-prefetchable) [size=4K] Bus: primary=09, secondary=0a, subordinate=0d, sec-latency=176 Memory window 0: 40000000-43fff000 (prefetchable) Memory window 1: 48000000-4bfff000 I/O window 0: 0000a400-0000a4ff I/O window 1: 0000a800-0000a8ff BridgeCtl: Parity- SERR- ISA- VGA- MAbort- >Reset+ 16bInt+ PostWrite+ 16-bit legacy interface ports at 0001 09:02.0 Ethernet controller [0200]: Realtek Semiconductor Co., Ltd. RTL-8139/8139C/8139C+ [10ec:8139] (rev 10) Subsystem: Toshiba America Info Systems Unknown device [1179:ff31] Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 64 (8000ns min, 16000ns max) Interrupt: pin A routed to IRQ 21 Region 0: I/O ports at a000 [size=256] Region 1: Memory at c0100000 (32-bit, non-prefetchable) [size=256] Capabilities: <access denied> 09:04.0 Ethernet controller [0200]: Atheros Communications, Inc. AR2413 802.11bg NIC [168c:001a] (rev 01) Subsystem: Askey Computer Corp. Unknown device [144f:7094] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV+ VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 168 (2500ns min, 7000ns max), Cache Line Size: 32 bytes Interrupt: pin A routed to IRQ 18 Region 0: Memory at c0110000 (32-bit, non-prefetchable) [size=64K] Capabilities: <access denied>

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[alsa-devel] [PATCH] speaker-test.c - fix pink noise generator on big-endian archs
by Giuliano Pochini 28 Apr '08

28 Apr '08

This patch makes speaker-test fill the buffers with properly coded data on both big- and little-endian processors. Signed-off-by: Giuliano Pochini <pochini(a)shiny.it> --- speaker-test/speaker-test.c__orig 2008-04-27 22:22:14.000000000 +0200 +++ speaker-test/speaker-test.c 2008-04-27 22:38:09.000000000 +0200 @@ -259,7 +259,8 @@ static void generate_pink_noise( uint8_t while (count-- > 0) { for(chn=0;chn<channels;chn++) { - if (sample_size_bits == 8) { + switch (format) { + case SND_PCM_FORMAT_S8: if (chn==channel) { res = generate_pink_noise_sample(&pink) * 0x03fffffff; /* Don't use MAX volume */ ires = res; @@ -267,35 +268,64 @@ static void generate_pink_noise( uint8_t } else { *samp8++ = 0; } - } else if (sample_size_bits == 16) { + break; + case SND_PCM_FORMAT_S16_LE: if (chn==channel) { res = generate_pink_noise_sample(&pink) * 0x03fffffff; /* Don't use MAX volume */ ires = res; - *samp16++ = ires >>16; +#if __BYTE_ORDER == __LITTLE_ENDIAN + *samp16++ = ires >> 16; +#elif __BYTE_ORDER == __BIG_ENDIAN + *samp16++ = bswap_16(ires >> 16); +#endif } else { *samp16++ = 0; } - } else if ((sample_size_bits == 32) && (format == SND_PCM_FORMAT_FLOAT_LE)) { + break; + case SND_PCM_FORMAT_S16_BE: if (chn==channel) { - res = generate_pink_noise_sample(&pink) * 0.75; /* Don't use MAX volume */ - fres = res; - *samp_f++ = fres; + res = generate_pink_noise_sample(&pink) * 0x03fffffff; /* Don't use MAX volume */ + ires = res; +#if __BYTE_ORDER == __BIG_ENDIAN + *samp16++ = ires >> 16; +#elif __BYTE_ORDER == __LITTLE_ENDIAN + *samp16++ = bswap_16(ires >> 16); +#endif } else { - *samp_f++ = 0.0; + *samp16++ = 0; } - } else if ((sample_size_bits == 32) && (format != SND_PCM_FORMAT_FLOAT_LE)) { + break; + case SND_PCM_FORMAT_S32_LE: + if (chn==channel) { + res = generate_pink_noise_sample(&pink) * 0x03fffffff; /* Don't use MAX volume */ + ires = res; +#if __BYTE_ORDER == __LITTLE_ENDIAN + *samp32++ = ires; +#elif __BYTE_ORDER == __BIG_ENDIAN + *samp32++ = bswap_32(ires); +#endif + } else { + *samp32++ = 0; + } + break; + case SND_PCM_FORMAT_S32_BE: if (chn==channel) { res = generate_pink_noise_sample(&pink) * 0x03fffffff; /* Don't use MAX volume */ ires = res; +#if __BYTE_ORDER == __BIG_ENDIAN *samp32++ = ires; +#elif __BYTE_ORDER == __LITTLE_ENDIAN + *samp32++ = bswap_32(ires); +#endif } else { *samp32++ = 0; } + break; + default: + ; } } - } - } static int set_hwparams(snd_pcm_t *handle, snd_pcm_hw_params_t *params, snd_pcm_access_t access) { -- Giuliano.

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[alsa-devel] No sound/mixer with new laptops on board sound - hda_intel ALC883
by Kristoffer Janke 27 Apr '08

27 Apr '08

Hello list, I have a newer Medion Akoya MD 96420 Laptop (Mitac 8227D Barebone). The card is an Intel HDA compatible device, the codec is ALC883. There is no suitable model for the alc883 codec, tested all - no sound. So I think there is some work needed, I have not much knowledge in this area but would like to help if possible. Alsa-info.sh output with newest(hg 6120) alsa driver: http://pastebin.ca/997178 -- Kristoffer Janke

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[alsa-devel] What is EAPD ?
by Roman Rybalko 27 Apr '08

27 Apr '08

Hi everyone, Does anybody know what is this "EAPD" ? Should I enable it or not ? What does it stands for ?

3 2

[alsa-devel] How can you tell if the speaker jack on your sound card has anything plugged into it???
by Sean Thayne 26 Apr '08

26 Apr '08

Hi Everybody, I was talking with some users in the irc channel in the free node, and was hoping to find a way to get the information about weither or not anything was plugged into the sound card, I've seen this feature with windows vista. I search thru a bunch of documentation sites and also scanned the /proc & /sys to no avail... Thanks, Sean T

6 11

[PATCH] Dirty hack to workaround pxa27x buggy hardreset.
by Robert Jarzmik 25 Apr '08

25 Apr '08

Signed-off-by: Robert Jarzmik <rjarzmik(a)free.fr> --- sound/soc/codecs/wm9713.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 2cb4fb8..8ec7ce9 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -975,6 +975,7 @@ static int wm9713_resume(struct platform_device *pdev) u16 id; /* give the codec an AC97 warm reset to start the link */ + codec->ac97->bus->ops->reset(codec->ac97); codec->ac97->bus->ops->warm_reset(codec->ac97); id = codec->soc_phys_read(codec->ac97, AC97_VENDOR_ID2); if (id != 0x4c13) { -- 1.5.5.3 --=-=-= Content-Type: text/x-diff Content-Disposition: inline; filename=0003-Added-double-ac97-reset-in-pxa27x-resume-path.patch

1 0

[alsa-devel] How can you tell if the speaker jack on your sound card has anything plugged into it???
by Sean Thayne 25 Apr '08

25 Apr '08

Hi Everybody, I was talking with some users in the irc channel in the free node, and was hoping to find a way to get the information about weither or not anything was plugged into the sound card, I've seen this feature with windows vista. I search thru a bunch of documentation sites and also scanned the /proc & /sys to no avail... Thanks, Sean T

1 0

[alsa-devel] How can you tell if the speaker jack on your sound card has anything plugged into it???
by Sean Thayne 25 Apr '08

25 Apr '08

Hi Everybody, I was talking with some users in the irc channel in the free node, and was hoping to find a way to get the information about weither or not anything was plugged into the sound card, I've seen this feature with windows vista. I search thru a bunch of documentation sites and also scanned the /proc & /sys to no avail... Thanks, Sean T

1 0

[alsa-devel] Status of X-Fi support in ALSA?
by James Pearson 25 Apr '08

25 Apr '08

Out of general interest, what is the current status of X-Fi support? On a related note, I notice that Creative have recently released the 2nd beta of their closed source drivers: <http://connect.creativelabs.com/opensource/Wiki/SoundCard%20Support.aspx> Thanks James Pearson

2 1

[alsa-devel] [RFC] ucb1400 touchscreen, irq auto probing and ac97 with its private field
by Sebastian Siewior 25 Apr '08

25 Apr '08

Hello, I've sent a RFC to the alsa mailing list [1] about adding an extra field in order to pass the IRQ from the AC97 driver to the ucb1400 driver. The result was: - William Pitcock recommended to hack up the ucb1400 driver and save the irq there. The result is [2]. What I don't like very much is the global variable and the fact that has to be done for every platform. - Mark Brown has the same problem with his wm97xx driver [3]. and while browsing a dead list archive of linux-input I noticed that a guy named Peter Ma has the same problem on his AVR32 [4]. I haven't heard anything from the ALSA Maintainer but according to git he didn't put a sing-of-by line since Thu Jan 31 17:40:18 2008 +0100 so maybe he is sleeping. That's why I've add Takashi Iwai since he seems to be in charge some how. Now I'm curious what solution the people here prefer: - adding a private field [1] (my favorite) - hacking up the ucb1400 [2] (doesn't solve [3] and needs addition code to solve [4]). - something totally different what did not come to my attention yet. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2008-March/006555.html [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2008-March/006559.html [3] http://mailman.alsa-project.org/pipermail/alsa-devel/2008-March/006564.html [4] http://www.mail-archive.com/linux-input@vger.kernel.org/msg00307.html Sebastian

5 32

[alsa-devel] [PATCH 1/3] asoc tlv320aic3x: revisit clock setup
by Daniel Mack 25 Apr '08

25 Apr '08

This patch cleans up the clocking setup for aic3x codecs. It drops the dividers table and determines the PLL control values programatically. Under certain conditions, the PLL is disabled entirely which could save some power. Signed-off-by: Daniel Mack <daniel(a)caiaq.de>

3 10

[alsa-devel] [PATCH 3/3] asoc tlv320aic3x: add more routing controls
by Daniel Mack 25 Apr '08

25 Apr '08

Add more routing controls to AIC3x chips to allow things like routing the left PGA input to right line out. Signed-off-by: Daniel Mack <daniel(a)caiaq.de>

5 7

[alsa-devel] [PATCH 2/3] asoc tlv320aic3x: add GPIO support
by Daniel Mack 25 Apr '08

25 Apr '08

This patch adds support for AIC3x GPIO lines. They can be configured for many possible functions as well as be driven manually. I also introduced i2c read functionality since the GPIO state register has to be read from hardware every time and can not be served from cache. Signed-off-by: Daniel Mack <daniel(a)caiaq.de>

2 3

[alsa-devel] asoc tsc210x
by ext-jukka.hynninen＠vaisala.com 25 Apr '08

25 Apr '08

Hello all, I'm assessing TSC2100 codec on an embdedded system. Any intentions on implementing an ASoC codec driver for TSC210x family? Could the multi-function nature of the chip raise some issues with ASoC framework? -Jukka

3 2

[alsa-devel] intel8x0 - copy function
by Harsha priya gupta 25 Apr '08

25 Apr '08

Hi, I am using Intel8x0 driver for my sound card. During data transfer from user to driver, I am trying to collect the buffer. I have implemented .copy function and also disabled the mmap. If I disable the mmap in .info structure, i get an error that the device is in use. If i try getting the debug messages; It fails in snd_pcm_common_ioctl1 for SNDRV_PCM_IOCTL_HW_REFINE. Can anyone please guide me on to the right path of implementing what I want to do. -- -Harsha

2 3

[alsa-devel] [RFC] [PATCH] asoc tlv320aic3x: revisit clocking setup
by Daniel Mack 25 Apr '08

25 Apr '08

Hi, attached is a patch that reworks the clocking setup for the aic3x codec. It drops the dividers table and calculates the values instead. When simulated, this code does the right thing for all the examples described in the datasheet. Please review and let me know about any concerns. BTW - I can't get in touch with the original author of this driver as his mail address doesn't work: <vbarinov(a)ru.mvista.com>: host mx2.mvista.com[206.112.117.9] said: 451 qq internal bug (#4.3.0) (in reply to end of DATA command) Does anyone have another contact? Best regards, Daniel

2 3

[alsa-devel] [RFC / PATCH] [SOUND] add a private field for ac97-device drivers and let ucb1400 be its first user
by Sebastian Siewior 25 Apr '08

25 Apr '08

From: Sebastian Siewior <bigeasy(a)linutronix.de> Currently the UCB1400 driver discovers the interrupt via probing. This works probably only on x86. This patch adds a private field to the ac97 struct where the ac97 driver can deposit informations for the device driver that serves a device which is attached to the ac97 bus. This patch also converts the UCB1400 driver to use this information if available. Signed-off-by: Sebastian Siewior <bigeasy(a)linutronix.de> --- drivers/input/touchscreen/ucb1400_ts.c | 12 ++++++++---- include/sound/ac97_codec.h | 4 ++++ 2 files changed, 12 insertions(+), 4 deletions(-) diff --git a/drivers/input/touchscreen/ucb1400_ts.c b/drivers/input/touchscreen/ucb1400_ts.c index 5ee175e..531fb4b 100644 --- a/drivers/input/touchscreen/ucb1400_ts.c +++ b/drivers/input/touchscreen/ucb1400_ts.c @@ -494,10 +494,14 @@ static int ucb1400_ts_probe(struct device *dev) goto err_free_devs; } - error = ucb1400_detect_irq(ucb); - if (error) { - printk(KERN_ERR "UCB1400: IRQ probe failed\n"); - goto err_free_devs; + if (!ucb->ac97->device_private_data) { + error = ucb1400_detect_irq(ucb); + if (error) { + printk(KERN_ERR "UCB1400: IRQ probe failed\n"); + goto err_free_devs; + } + } else { + ucb->irq = (int) ucb->ac97->device_private_data; } error = request_irq(ucb->irq, ucb1400_hard_irq, IRQF_TRIGGER_RISING, diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index 0148058..b1af896 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -474,6 +474,10 @@ struct snd_ac97 { struct snd_ac97_build_ops * build_ops; void *private_data; void (*private_free) (struct snd_ac97 *ac97); + /* This field is used by device drivers which serve devices which are + * attached to the AC97 bus. + */ + void *device_private_data; /* --- */ struct snd_ac97_bus *bus; struct pci_dev *pci; /* assigned PCI device - used for quirks */ -- 1.5.4.3

2 2

[alsa-devel] Lenovo 3000 N100 ALC861-VD quirk
by Chris Pemberton 25 Apr '08

25 Apr '08

Hello, I'm revisiting an unresolved bug report I opened several months ago: bug # 0003468 The bug still exists as of ALSA v1.0.15 (Ubuntu 8.04). A quick recap: The following line in patch_realtek.c is my machine: SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo", ALC861VD_LENOVO), If I edit this line as follows, I get sound: SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo" ALC861VD_3ST), (or I can just pass model=3stack to get sound). Inserting headphones doesn't mute the speakers or power the headphones. But I can manually unmute the headphones via alsamixer (but speakers are still not muted). I do see that the special keyboard keys for speaker mute, volume up and volume down are working correctly; but I don't know if this is something ALSA is responsible for, or X, or Ubuntu. So if this is ALSA, thanks! Anything more I can do to help resolve this? Thanks, Chris

2 4

[alsa-devel] 2.6.25-rc8: why #define SND_HDA_NEEDS_RESUME is not set ?
by warped 25 Apr '08

25 Apr '08

Hi all, I have issue with 2.6.24/25 kernels and analog outs after s3 resume. Simply: after s3 resume analog outs are dead (no sound). SPDIF works OK. With 2.6.23.xx all was OK I browse ALSA code in 2.6.25-rc8 and see that in hda_codec.h define SND_HDA_NEEDS_RESUME is commented. Q is: -might uncommenting this line help me with my issue ? -why it is commented ? My hw is Abit AN-M2HD with ALC889. During boot Kernel reports: user.info kernel: hda_codec: Unknown model for ALC883, trying auto-probe from BIOS...

3 4

Re: [alsa-devel] [alsa-cvslog] alsa-kernel: add a private field for ac97-device drivers and let ucb1400 be its first user
by Takashi Iwai 25 Apr '08

25 Apr '08

At Fri, 25 Apr 2008 08:29:08 +0200 (CEST), Jaroslav Kysela wrote: > > changeset: 6120:e2ff47e8771b > tag: tip > user: perex > date: Fri Apr 25 08:29:05 2008 +0200 > files: include/ac97_codec.h kernel/drivers/input/touchscreen/ucb1400_ts.c > description: > add a private field for ac97-device drivers and let ucb1400 be its first user > From: Sebastian Siewior <bigeasy(a)linutronix.de> > > Currently the UCB1400 driver discovers the interrupt via probing. This works > probably only on x86. This patch adds a private field to the ac97 struct > where the ac97 driver can deposit informations for the device driver that > serves a device which is attached to the ac97 bus. > This patch also converts the UCB1400 driver to use this information if > available. > > Signed-off-by: Sebastian Siewior <bigeasy(a)linutronix.de> Jaroslav, as I posted yesterday, I don't think it's good to use a void pointer blindly there. In this case, it's different from a normal private_data, which is asigned and used by the same driver. This is a data passed by ac97 controller driver to another device driver. Thus, it should be a strong type. Also, a cast from a pointer to an int causes a compile warning. So, could you revert it until all issues are solved? thanks, Takashi > > diff -r 9c74d781f388 -r e2ff47e8771b include/ac97_codec.h > --- a/include/ac97_codec.h Fri Apr 25 08:15:12 2008 +0200 > +++ b/include/ac97_codec.h Fri Apr 25 08:29:05 2008 +0200 > @@ -475,6 +475,10 @@ struct snd_ac97 { > struct snd_ac97_build_ops * build_ops; > void *private_data; > void (*private_free) (struct snd_ac97 *ac97); > + /* This field is used by device drivers which serve devices which are > + * attached to the AC97 bus. > + */ > + void *device_private_data; > /* --- */ > struct snd_ac97_bus *bus; > struct pci_dev *pci; /* assigned PCI device - used for quirks */ > diff -r 9c74d781f388 -r e2ff47e8771b kernel/drivers/input/touchscreen/ucb1400_ts.c > --- a/kernel/drivers/input/touchscreen/ucb1400_ts.c Fri Apr 25 08:15:12 2008 +0200 > +++ b/kernel/drivers/input/touchscreen/ucb1400_ts.c Fri Apr 25 08:29:05 2008 +0200 > @@ -492,10 +492,14 @@ static int ucb1400_ts_probe(struct devic > goto err_free_devs; > } > > - error = ucb1400_detect_irq(ucb); > - if (error) { > - printk(KERN_ERR "UCB1400: IRQ probe failed\n"); > - goto err_free_devs; > + if (!ucb->ac97->device_private_data) { > + error = ucb1400_detect_irq(ucb); > + if (error) { > + printk(KERN_ERR "UCB1400: IRQ probe failed\n"); > + goto err_free_devs; > + } > + } else { > + ucb->irq = (int) ucb->ac97->device_private_data; > } > > error = request_irq(ucb->irq, ucb1400_hard_irq, IRQF_TRIGGER_RISING, > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by the 2008 JavaOne(SM) Conference > Don't miss this year's exciting event. There's still time to save $100. > Use priority code J8TL2D2. > http://ad.doubleclick.net/clk;198757673;13503038;p?http://java.sun.com/java… > _______________________________________________ > Alsa-cvslog mailing list > Alsa-cvslog(a)lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/alsa-cvslog >

1 0

[alsa-devel] [PATCH v2][ASoC v2] Replace snd_soc_free_codec with snd_soc_unregister_codec
by Timur Tabi 24 Apr '08

24 Apr '08

The functions snd_soc_register_codec() and snd_soc_free_codec() are companion functions, but snd_soc_free_codec() releases resources (via kfree) that were not allocated by snd_soc_register_codec(). This means that the caller of snd_soc_register_codec() has to use a specific method to allocate these resources. So we replace snd_soc_free_codec() with snd_soc_unregister_codec(). This new function only unregisters the codec but does not free the memory resources. The caller can now allocate and free the memory resources as it sees fit. Also updated all callers of snd_soc_free_codec() to call snd_soc_unregister_codec() instead and free the resources themselves. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- I cannot get ASoC V2 to compile on my machine, so this code is untested. include/sound/soc-codec.h | 2 +- sound/soc/codecs/wm8350.c | 8 ++++++-- sound/soc/codecs/wm8731.c | 8 ++++++-- sound/soc/codecs/wm8750.c | 8 ++++++-- sound/soc/codecs/wm8753.c | 8 ++++++-- sound/soc/codecs/wm9712.c | 8 ++++++-- sound/soc/codecs/wm9713.c | 8 ++++++-- sound/soc/soc-core.c | 11 ++++------- 8 files changed, 41 insertions(+), 20 deletions(-) diff --git a/include/sound/soc-codec.h b/include/sound/soc-codec.h index 3cf4819..a809160 100644 --- a/include/sound/soc-codec.h +++ b/include/sound/soc-codec.h @@ -160,7 +160,7 @@ struct snd_soc_codec *snd_soc_new_codec( int snd_soc_register_codec(struct snd_soc_codec *codec, struct device *dev); -void snd_soc_free_codec(struct snd_soc_codec *codec); +void snd_soc_unregister_codec(struct snd_soc_codec *codec); int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, struct snd_ac97_bus_ops *ops, struct snd_card *card, diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 3195d13..4ea8ced 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1436,7 +1436,9 @@ static int wm8350_codec_probe(struct platform_device *pdev) codec_err: kfree(wm8350); prv_err: - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return ret; } @@ -1446,7 +1448,9 @@ static int wm8350_codec_remove(struct platform_device *pdev) struct wm8350_data *wm8350 = codec->private_data; snd_soc_unregister_codec_dai(wm8350->dai); - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); kfree(wm8350); return 0; } diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 0fe8766..2195c77 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -595,7 +595,9 @@ static int wm8731_codec_probe(struct platform_device *pdev) codec_err: kfree(wm8731); wm8731_err: - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return ret; } @@ -606,7 +608,9 @@ static int wm8731_codec_remove(struct platform_device *pdev) snd_soc_unregister_codec_dai(wm8731->dai); kfree(wm8731); - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return 0; } diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 1b0296f..23e9126 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -898,7 +898,9 @@ static int wm8750_codec_probe(struct platform_device *pdev) codec_err: kfree(wm8750); wm8750_err: - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return ret; } @@ -909,7 +911,9 @@ static int wm8750_codec_remove(struct platform_device *pdev) snd_soc_unregister_codec_dai(wm8750->dai); kfree(wm8750); - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return 0; } static struct platform_driver wm8750_codec_driver = { diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 0d795f4..1cb2d64 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1584,7 +1584,9 @@ voice_err: codec_err: kfree(codec->private_data); err: - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return ret; } @@ -1603,7 +1605,9 @@ static int wm8753_codec_remove(struct platform_device *pdev) /* free codec */ kfree(codec->private_data); - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return 0; } diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 16b6a18..7014190 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -715,7 +715,9 @@ aux_dai_err: codec_err: kfree(wm9712); priv_err: - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return ret; } @@ -728,7 +730,9 @@ static int wm9712_codec_remove(struct platform_device *pdev) snd_soc_unregister_codec_dai(wm9712->aux_dai); kfree(codec->private_data); - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return 0; } diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 5e99bbe..d9ded51 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1154,7 +1154,9 @@ aux_dai_err: codec_err: kfree(wm9713); priv_err: - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return ret; } @@ -1168,7 +1170,9 @@ static int wm9713_codec_remove(struct platform_device *pdev) snd_soc_unregister_codec_dai(wm9713->voice_dai); kfree(codec->private_data); - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return 0; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 939b543..7dc6d68 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1856,21 +1856,18 @@ int snd_soc_register_codec(struct snd_soc_codec *codec, struct device *dev) EXPORT_SYMBOL_GPL(snd_soc_register_codec); /** - * snd_soc_free_codec - unregister and free codec. + * snd_soc_unregister_codec - unregister a codec. * @codec: codec driver * - * Unregisters a codec driver with the core and frees all its resources. + * Unregisters a codec driver with the core. */ -void snd_soc_free_codec(struct snd_soc_codec *codec) +void snd_soc_unregister_codec(struct snd_soc_codec *codec) { mutex_lock(&client_mutex); list_del(&codec->list); mutex_unlock(&client_mutex); - if (codec->reg_cache) - kfree(codec->reg_cache); - kfree(codec); } -EXPORT_SYMBOL_GPL(snd_soc_free_codec); +EXPORT_SYMBOL_GPL(snd_soc_unregister_codec); /** * snd_soc_register_platform_dai - registers a platform DAI. -- 1.5.5.1

2 1

[alsa-devel] [PATCH][ASoC v2] Replace snd_soc_free_codec with snd_soc_unregister_codec
by Timur Tabi 24 Apr '08

24 Apr '08

The functions snd_soc_register_codec() and snd_soc_free_codec() are companion functions, but snd_soc_free_codec() releases resources (via kfree) that were not allocated by snd_soc_register_codec(). This means that the caller of snd_soc_register_codec() has to use a specific method to allocate these resources. So we replace snd_soc_free_codec() with snd_soc_unregister_codec(). This new function only unregisters the codec but does not free the memory resources. The caller can now allocate and free the memory resources as it sees fit. Also updated all callers of snd_soc_free_codec() to call snd_soc_unregister_codec() instead and free the resources themselves. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- include/sound/soc-codec.h | 2 +- sound/soc/codecs/wm8350.c | 8 ++++++-- sound/soc/codecs/wm8731.c | 8 ++++++-- sound/soc/codecs/wm8750.c | 8 ++++++-- sound/soc/codecs/wm8753.c | 8 ++++++-- sound/soc/codecs/wm9712.c | 8 ++++++-- sound/soc/codecs/wm9713.c | 8 ++++++-- sound/soc/soc-core.c | 11 ++++------- 8 files changed, 41 insertions(+), 20 deletions(-) diff --git a/include/sound/soc-codec.h b/include/sound/soc-codec.h index 3cf4819..6f132ba 100644 --- a/include/sound/soc-codec.h +++ b/include/sound/soc-codec.h @@ -160,7 +160,7 @@ struct snd_soc_codec *snd_soc_new_codec( int snd_soc_register_codec(struct snd_soc_codec *codec, struct device *dev); -void snd_soc_free_codec(struct snd_soc_codec *codec); +void snd_soc_deregister_codec(struct snd_soc_codec *codec); int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, struct snd_ac97_bus_ops *ops, struct snd_card *card, diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 3195d13..4ea8ced 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1436,7 +1436,9 @@ static int wm8350_codec_probe(struct platform_device *pdev) codec_err: kfree(wm8350); prv_err: - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return ret; } @@ -1446,7 +1448,9 @@ static int wm8350_codec_remove(struct platform_device *pdev) struct wm8350_data *wm8350 = codec->private_data; snd_soc_unregister_codec_dai(wm8350->dai); - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); kfree(wm8350); return 0; } diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 0fe8766..2195c77 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -595,7 +595,9 @@ static int wm8731_codec_probe(struct platform_device *pdev) codec_err: kfree(wm8731); wm8731_err: - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return ret; } @@ -606,7 +608,9 @@ static int wm8731_codec_remove(struct platform_device *pdev) snd_soc_unregister_codec_dai(wm8731->dai); kfree(wm8731); - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return 0; } diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 1b0296f..23e9126 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -898,7 +898,9 @@ static int wm8750_codec_probe(struct platform_device *pdev) codec_err: kfree(wm8750); wm8750_err: - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return ret; } @@ -909,7 +911,9 @@ static int wm8750_codec_remove(struct platform_device *pdev) snd_soc_unregister_codec_dai(wm8750->dai); kfree(wm8750); - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return 0; } static struct platform_driver wm8750_codec_driver = { diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 0d795f4..1cb2d64 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1584,7 +1584,9 @@ voice_err: codec_err: kfree(codec->private_data); err: - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return ret; } @@ -1603,7 +1605,9 @@ static int wm8753_codec_remove(struct platform_device *pdev) /* free codec */ kfree(codec->private_data); - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return 0; } diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 16b6a18..7014190 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -715,7 +715,9 @@ aux_dai_err: codec_err: kfree(wm9712); priv_err: - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return ret; } @@ -728,7 +730,9 @@ static int wm9712_codec_remove(struct platform_device *pdev) snd_soc_unregister_codec_dai(wm9712->aux_dai); kfree(codec->private_data); - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return 0; } diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 5e99bbe..d9ded51 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1154,7 +1154,9 @@ aux_dai_err: codec_err: kfree(wm9713); priv_err: - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return ret; } @@ -1168,7 +1170,9 @@ static int wm9713_codec_remove(struct platform_device *pdev) snd_soc_unregister_codec_dai(wm9713->voice_dai); kfree(codec->private_data); - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return 0; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 939b543..7dc6d68 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1856,21 +1856,18 @@ int snd_soc_register_codec(struct snd_soc_codec *codec, struct device *dev) EXPORT_SYMBOL_GPL(snd_soc_register_codec); /** - * snd_soc_free_codec - unregister and free codec. + * snd_soc_unregister_codec - unregister a codec. * @codec: codec driver * - * Unregisters a codec driver with the core and frees all its resources. + * Unregisters a codec driver with the core. */ -void snd_soc_free_codec(struct snd_soc_codec *codec) +void snd_soc_unregister_codec(struct snd_soc_codec *codec) { mutex_lock(&client_mutex); list_del(&codec->list); mutex_unlock(&client_mutex); - if (codec->reg_cache) - kfree(codec->reg_cache); - kfree(codec); } -EXPORT_SYMBOL_GPL(snd_soc_free_codec); +EXPORT_SYMBOL_GPL(snd_soc_unregister_codec); /** * snd_soc_register_platform_dai - registers a platform DAI. -- 1.5.5.1

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Re: [alsa-devel] how to disable mmap and allow .copy function to be called
by Harsha priya gupta 24 Apr '08

24 Apr '08

Thank you. I tried that but on starting a playback it gives me an error that the device is in use and does not playback. Should i be enabling anything else for my .copy function to be invoked and successful playback to happen? On Thu, Apr 24, 2008 at 10:53 AM, Eliot Blennerhassett < linux(a)audioscience.com> wrote: > Harsha priya gupta wrote: > > Hi, > > > > Can anyone help me - I need to modify an existing driver such that a new > > .copy function is invoked. how to make sure that mmap is diabled? > > > > Remove SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID > > from the struct snd_pcm_hardware.info initialisation. > > -- > Eliot > > > > > -- -Harsha

1 0

[alsa-devel] how to disable mmap and allow .copy function to be called
by Harsha priya gupta 24 Apr '08

24 Apr '08

Hi, Can anyone help me - I need to modify an existing driver such that a new .copy function is invoked. how to make sure that mmap is diabled? Thanks, Harsha

1 0

[alsa-devel] snd_pcm_rewind() weirdness
by Lennart Poettering 24 Apr '08

24 Apr '08

Hey! On ALSA 1.0.16 with the HDA driver I am experiencing some weird things with snd_pcm_rewind() which I think are the result of a kernel bug, but I am not sure. Here's what I do to trigger this: I play some PCM. Eventually I want to rewrite what I already have written. I call snd_pcm_hwsync() and then snd_pcm_avail_update() to determine how "empty" the playback buffer is. I subtract that value from the total buffer size. The resulting value should be something resembling the number of frames that I might be able to rewind. I then subtract enough frames to keep away from the playback pointer far enough (50ms, the playback buffer is 370ms, period is 185ms). Now I pass that value to snd_pcm_rewind() and it returns something that is higher than what I passed in. Which I find quite surprising. I had expected that the returned value could be equal or smaller than what I passed in, but never higher! So, my interpretation of the API was apparently wrong. It looks like on HDA the snd_pcm_rewind() parameter is apparently rounded *up* to the next period boundary (which in this case was equivalent to the *entire* buffer)? Of course, rewinding that far immediately caused an XRUN. Which is why I think rounding up in this cases is a bad idea and the API shouldn't work this way. Rounding down is OK, rounding up is call for trouble. To tape over this I added some code to immediately call snd_pcm_forward() with the difference if I noticed that the device forwarded more than I requested. This value also got rounded up -- to twice the period size. So with this backward+forward operation I ended up at a frame index that is farther ahead than where I originally came from! So, what's the expected behaviour of snd_pcm_rewind()? Always round up? What's the rationale behind this? If this is not a bug but expected behaviour, could we at least add something for passing the direction to round to? What I am working on is the "glitch-free" playback model in PulseAudio (http://0pointer.de/blog/projects/pulse-glitch-free.html). To do this I am writing blocks smaller than the period size to the device via the mmap interface all the time. Given that on HDA _rewind() is apparently not granular to the frame size, is writing via mmap in small blocks like that OK? If it's not, how can I learn about this in advance? 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] [RFC/PATCH 0/1] ALSA SoC support for OMAP
by Jarkko Nikula 24 Apr '08

24 Apr '08

Hi Here is the second version of ASoC support for OMAP. Now based to dev branch of git://opensource.wolfsonmicro.com/linux-2.6-asoc but applies to linux-omap by modifying Kconfig and Makefile in sound/soc/. Code hasn't changed a very much. Some minor clean-ups etc and support for OMAP2430 and 3430 is not included until we have updated arch/arm/plat-omap/mcbsp.c module in place supporting them (work in progress). This have the same issues than first version [1] but TODO list is different: - Pin multiplexing: This belongs to .../plat-omap/mcbsp.c, not here - Multiformat & multilink McBSP DAI: ASoC v2 - omap_mcbsp_start & _stop API change: Probably with or after updated mcbsp.c Note that include/asm-arm/arch-omap/control.h is missing from vanilla kernel at the moment but that's queued for 2.6.26. -- Jarkko 1. http://marc.info/?l=linux-omap&m=120643770810683&w=2

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[alsa-devel] [PATCH 01/11] soc - s3c24xx-i2s - Use linux/io.h
by Mark Brown 23 Apr '08

23 Apr '08

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/s3c24xx/s3c24xx-i2s.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 0de211c..c95ebe2 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -25,6 +25,7 @@ #include <linux/delay.h> #include <linux/clk.h> #include <linux/jiffies.h> +#include <linux/io.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -32,7 +33,6 @@ #include <sound/soc.h> #include <asm/hardware.h> -#include <asm/io.h> #include <asm/arch/regs-gpio.h> #include <asm/arch/regs-clock.h> #include <asm/arch/audio.h> -- 1.5.5.1

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[alsa-devel] [PATCH 1/5] soc - Support PXA3xx AC97
by Mark Brown 23 Apr '08

23 Apr '08

The PXA3xx does not support the use of interrupts during reset and access to the GPIO status requires similar handling to that for PXA27x. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/pxa/pxa2xx-ac97.c | 29 +++++++++++++++++++++++++++-- 1 files changed, 27 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 815c153..95ae509 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -55,7 +55,7 @@ static unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, mutex_lock(&car_mutex); /* set up primary or secondary codec/modem space */ -#ifdef CONFIG_PXA27x +#if defined(CONFIG_PXA27x) || defined(CONFIG_PXA3xx) reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE; #else if (reg == AC97_GPIO_STATUS) @@ -105,7 +105,7 @@ static void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, mutex_lock(&car_mutex); /* set up primary or secondary codec/modem space */ -#ifdef CONFIG_PXA27x +#if defined(CONFIG_PXA27x) || defined(CONFIG_PXA3xx) reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE; #else if (reg == AC97_GPIO_STATUS) @@ -128,6 +128,9 @@ static void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) { +#ifdef CONFIG_PXA3xx + int timeout = 100; +#endif gsr_bits = 0; #ifdef CONFIG_PXA27x @@ -138,6 +141,11 @@ static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) GCR |= GCR_WARM_RST; pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); udelay(500); +#elif defined(CONFIG_PXA3xx) + /* Can't use interrupts */ + GCR |= GCR_WARM_RST; + while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--) + mdelay(1); #else GCR |= GCR_WARM_RST | GCR_PRIRDY_IEN | GCR_SECRDY_IEN; wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1); @@ -153,6 +161,16 @@ static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97) { +#ifdef CONFIG_PXA3xx + int timeout = 1000; + + /* Hold CLKBPB for 100us */ + GCR = 0; + GCR = GCR_CLKBPB; + udelay(100); + GCR = 0; +#endif + GCR &= GCR_COLD_RST; /* clear everything but nCRST */ GCR &= ~GCR_COLD_RST; /* then assert nCRST */ @@ -164,6 +182,13 @@ static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97) pxa_set_cken(CKEN_AC97CONF, 0); GCR = GCR_COLD_RST; udelay(50); +#elif defined(CONFIG_PXA3xx) + /* Can't use interrupts on PXA3xx */ + GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN); + + GCR = GCR_WARM_RST | GCR_COLD_RST; + while (!(GSR & (GSR_PCR | GSR_SCR)) && timeout--) + mdelay(10); #else GCR = GCR_COLD_RST; GCR |= GCR_CDONE_IE|GCR_SDONE_IE; -- 1.5.5

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[alsa-devel] Detecting when user is calling himself
by Mathieu Monney 23 Apr '08

23 Apr '08

Hi all I'm trying to figure out how to detect if the user is trying to call himself because it is making some weird unwanted things on my UI when user do this. Any idea? Thank you. Regards, Mathieu Monney

2 1

[alsa-devel] [PATCH 1/3] soc - s3c24xx-i2s - Replace __FUNCTION__ with __func__
by Mark Brown 22 Apr '08

22 Apr '08

__FUNCTION__ is GCC specific and checkpatch won't let us submit with it. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/s3c24xx/s3c24xx-i2s.c | 20 ++++++++++---------- 1 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 6b3cbbf..7d38f55 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -89,7 +89,7 @@ static void s3c24xx_snd_txctrl(int on) u32 iiscon; u32 iismod; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); @@ -134,7 +134,7 @@ static void s3c24xx_snd_rxctrl(int on) u32 iiscon; u32 iismod; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); @@ -182,7 +182,7 @@ static int s3c24xx_snd_lrsync(void) u32 iiscon; int timeout = 50; /* 5ms */ - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); while (1) { iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); @@ -202,7 +202,7 @@ static int s3c24xx_snd_lrsync(void) */ static inline int s3c24xx_snd_is_clkmaster(void) { - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); return (readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & S3C2410_IISMOD_SLAVE) ? 0:1; } @@ -215,7 +215,7 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, { u32 iismod; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); DBG("hw_params r: IISMOD: %lx \n", iismod); @@ -251,7 +251,7 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; u32 iismod; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) rtd->dai->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_out; @@ -279,7 +279,7 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd) { int ret = 0; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -321,7 +321,7 @@ static int s3c24xx_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, { u32 iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); iismod &= ~S3C2440_IISMOD_MPLL; @@ -347,7 +347,7 @@ static int s3c24xx_i2s_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai, { u32 reg; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); switch (div_id) { case S3C24XX_DIV_BCLK: @@ -382,7 +382,7 @@ EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate); static int s3c24xx_i2s_probe(struct platform_device *pdev) { - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); s3c24xx_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100); if (s3c24xx_i2s.regs == NULL) -- 1.5.5

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[alsa-devel] envy24control can't set sample rate on Delta 1010
by John Rigg 22 Apr '08

22 Apr '08

envy24control in alsa-tools-1.0.16 cannot set sample rate on the M-Audio Delta 1010. The Actual Rate window appears to be functioning correctly, as is the "S/PDIF in" button, but none of the internal clock sample rate buttons do anything. Setting the sample rate as a jackd parameter works as normal. I tried it with 2.6.20 and 2.6.25 kernels. Behaviour was the same on both. I've made a bug report (0003915). John

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Re: [alsa-devel] [PATCH] asoc tlv320aic3x: add GPIO support
by Daniel Mack 22 Apr '08

22 Apr '08

On Fri, Apr 18, 2008 at 11:26:28PM +0300, Jarkko Nikula wrote: > No any show stopper either but I just started to think about gpiolib if > AIC3x GPIOs are used as a GPIO. I agree in general and thought about alternatives as well, but in this case, it turns out to be a thing the aic3x driver is responsible to as the registers in questions are purely in its scope. So the idea is to set up the GPIO's functions via alsa controls and handle its events in some board-related code as this is really not abstractable. One could, however, do this setup in machine-related code within the asoc environment. My changes don't cast any problems on the existing implementation either, so I would suggest to just take this patch until someone comes up with a more suitable solution :) Best regards, Daniel

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[alsa-devel] [PATCH] ALSA: Fix capturing for envy1724ht based "audiophile 192"
by Karsten Wiese 22 Apr '08

22 Apr '08

The 1st hunk changes snd_ice1712_gpio_write_bits() to not affect the gpio's mask register. Lets gpio writers that are unaware of a changed mask register successfully write the gpio. The 2nd and 3rd hunks toggle "chip reset" and "eeprom based setup" sequence so "chip reset" becomes first. Increasement of the "chip reset" related timeout leads to correctly read eeprom's contents here. Also corrects some arguments in calls to snd_ice1712_gpio_write_bits() from ap192_set_rate_val(). Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de> --- Hi, patch might hamper cards that actually relied on snd_ice1712_gpio_write_bits() setting the gpio's mask register. Cscope tells me terratec ews and m-audio rev51 are might be candidates. I guess they are ok ;-) regards, Karsten diff -pur alsa.hg/alsa-kernel/pci/ice1712/ice1712.h fixed/alsa-kernel/pci/ice1712/ice1712.h --- alsa.hg/alsa-kernel/pci/ice1712/ice1712.h 2008-03-20 12:25:01.000000000 +0100 +++ fixed/alsa-kernel/pci/ice1712/ice1712.h 2008-04-19 23:21:31.000000000 +0200 @@ -438,10 +438,14 @@ int snd_ice1712_gpio_put(struct snd_kcon static inline void snd_ice1712_gpio_write_bits(struct snd_ice1712 *ice, unsigned int mask, unsigned int bits) { + unsigned val; + ice->gpio.direction |= mask; snd_ice1712_gpio_set_dir(ice, ice->gpio.direction); - snd_ice1712_gpio_set_mask(ice, ~mask); - snd_ice1712_gpio_write(ice, mask & bits); + val = snd_ice1712_gpio_read(ice); + val &= ~mask; + val |= mask & bits; + snd_ice1712_gpio_write(ice, val); } static inline int snd_ice1712_gpio_read_bits(struct snd_ice1712 *ice, diff -pur alsa.hg/alsa-kernel/pci/ice1712/ice1724.c fixed/alsa-kernel/pci/ice1712/ice1724.c --- alsa.hg/alsa-kernel/pci/ice1712/ice1724.c 2008-03-20 12:25:01.000000000 +0100 +++ fixed/alsa-kernel/pci/ice1712/ice1724.c 2008-04-19 23:07:29.000000000 +0200 @@ -2045,12 +2045,16 @@ static int __devinit snd_vt1724_read_eep -static int __devinit snd_vt1724_chip_init(struct snd_ice1712 *ice) +static void __devinit snd_vt1724_chip_reset(struct snd_ice1712 *ice) { outb(VT1724_RESET , ICEREG1724(ice, CONTROL)); - udelay(200); + msleep(10); outb(0, ICEREG1724(ice, CONTROL)); - udelay(200); + msleep(10); +} + +static int __devinit snd_vt1724_chip_init(struct snd_ice1712 *ice) +{ outb(ice->eeprom.data[ICE_EEP2_SYSCONF], ICEREG1724(ice, SYS_CFG)); outb(ice->eeprom.data[ICE_EEP2_ACLINK], ICEREG1724(ice, AC97_CFG)); outb(ice->eeprom.data[ICE_EEP2_I2S], ICEREG1724(ice, I2S_FEATURES)); @@ -2223,6 +2227,7 @@ static int __devinit snd_vt1724_create(s ice->irq = pci->irq; + snd_vt1724_chip_reset(ice); if (snd_vt1724_read_eeprom(ice, modelname) < 0) { snd_vt1724_free(ice); return -EIO; diff -pur alsa.hg/alsa-kernel/pci/ice1712/revo.c fixed/alsa-kernel/pci/ice1712/revo.c --- alsa.hg/alsa-kernel/pci/ice1712/revo.c 2008-03-17 09:25:02.000000000 +0100 +++ fixed/alsa-kernel/pci/ice1712/revo.c 2008-04-19 23:12:33.000000000 +0200 @@ -327,7 +327,7 @@ static void ap192_set_rate_val(struct sn revo_set_rate_val(ak, rate); /* reset CKS */ - snd_ice1712_gpio_write_bits(ice, 1 << 8, rate > 96000 ? 1 : 0); + snd_ice1712_gpio_write_bits(ice, 1 << 8, rate > 96000 ? 1 << 8 : 0); /* reset DFS pins of AK5385A for ADC, too */ if (rate > 96000) dfs = 2; @@ -338,7 +338,7 @@ static void ap192_set_rate_val(struct sn snd_ice1712_gpio_write_bits(ice, 3 << 9, dfs << 9); /* reset ADC */ snd_ice1712_gpio_write_bits(ice, 1 << 11, 0); - snd_ice1712_gpio_write_bits(ice, 1 << 11, 1); + snd_ice1712_gpio_write_bits(ice, 1 << 11, 1 << 11); } static const struct snd_akm4xxx_dac_channel ap192_dac[] = {

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Re: [alsa-devel] snd_pcm_ops - .copy
by Harsha priya gupta 22 Apr '08

22 Apr '08

On Tue, Apr 22, 2008 at 8:07 AM, Harsha priya gupta <harshapriya(a)gmail.com> wrote: > I am not using the mmap method. How can i make sure that everytime a user > buffer is passed to driver, my copy function would be called? > > > On Mon, Apr 21, 2008 at 9:30 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > At Mon, 21 Apr 2008 19:27:45 +0530, > > Harsha priya gupta wrote: > > > > > > Hi, > > > I am trying to implement the .copy function of playack_ops in my > > driver. But > > > when I try playing an mp3 file, my .copy's function code path was > > executing. > > > Suddenly I do not see that function being called. Can anyone help me > > > understand when the following code path will be executed > > > > > > snd_pcm_lib_write_transfer > > > substream->ops->copy > > > > Do you use a mmap method? If so, the copy callback won't be called. > > The copy callback is only for copying the data from the user-space > > buffer to the driver buffer, which is skipped via mmap. > > > > > > Takashi > > > > > > -- > -Harsha -- -Harsha

1 0

[alsa-devel] Alsa Callbacks and C++ classes
by Reuben Martin 22 Apr '08

22 Apr '08

Hello all, I have an AES32 sound card that I am trying to write a general control interface for using qt4. I'm wanting to register callbacks so that the interface will reflect changes made by other apps. Now I'm not entirely up to speed on the API, but from what I can gather, the callback can be registered to call a regular function, or to change a public class variable. Since I decided to use qt4, everything is in classes. Is there any way to have the callback trigger a public class function? I guess I don't understand why you would want to use a public variable instead. Wouldn't you have to set up a means within the class to continually pole the variable to see if it has been changed? Or can the option to call a regular function be used to call a public class function as well? Go easy on me. I'm not much of an experienced programmer. More of a hack who can kinda get by with his sloppy programming most of the time. Thanks, -Reuben

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[alsa-devel] Buffering issues
by Lennart Poettering 22 Apr '08

22 Apr '08

Heya! As you might now I am currently working on something called "glitch-free playback model" for PulseAudio: http://0pointer.de/blog/projects/pulse-glitch-free.html One of the central ideas is to schedule audio via timer interrupts, no longer via period-based interrupts. To acomplish this I need to be able to estimate in advance how long (in time) it will take until the playback buffer runs empty at a certain point. Then I can set myself a system timer to wakeup a certain margin earlier than this and fill the buffer up again. That way I minimize wakeups and should always have enough safety margin to never get an xrun. Now, the way I implemented this right now is: in each iteration of my playback loop I first fill up the playback buffer completely if it is not fully filled up yet. After that I obviously can go to sleep for the time it takes to play buffer_size samples minus some safety margin, which i set to 20ms. Apparently this is actually not as obvious as I thought. This works fine on many soundcards but it apparently doesn't work on USB hardware. There, ALSA detects a buffer underrun much earlier than when my timer elapses. My loop wakes up due to activity on the ALSA fd, not because of the timer. My code then goes and asks _avail_update() for the free space in the buffer and sees that the full buffer is available, thus assuming a xrun happened. This will cause PA to double the safety margin and do some other things to make sure an XRUN doesn't happen the next time again. So, how do I properly estimate how long the filled up buffer will take to play? Or to word it differently: how do I properly estimate how much time will pass until an XRUN would occur? [My interpretation of what's going on here with lots of guessing how things work is this: the buffer exposed by ALSA is merely a send buffer and just one of the buffers involved before the data actually reaches the speakers. ALSA just takes the data out of this buffer and sends it to the device via USB block-by-block. Thus the send buffer maintained by ALSA is much quicker empty than the audio data is actually played. And Boom! -- If that should actually be the case then I'd like to request an API that allows me to query an upper limit for how much buffering/latency actually happens after the audio data left the ALSA buffer. I could then use this an subtract it from my time to sleep.] Thanks, 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] snd_pcm_ops - .copy
by Harsha priya gupta 21 Apr '08

21 Apr '08

Hi, I am trying to implement the .copy function of playack_ops in my driver. But when I try playing an mp3 file, my .copy's function code path was executing. Suddenly I do not see that function being called. Can anyone help me understand when the following code path will be executed snd_pcm_lib_write_transfer substream->ops->copy -Harsha

2 1

[alsa-devel] Flash Plugin & IO Plug: Async interface not implemented?
by Alexander Indenbaum 21 Apr '08

21 Apr '08

Hello, I'm trying to integrate Firefox Flash plugin with ASLA IO plug driver for proprietary hardware. I've found out that Flash Plugin ASLA client side (sources could be found at http://labs.adobe.com/wiki/index.php/Flash_Player:Additional_Interface_Supp…) is incompatible with IO Plug. The reason is that during initialization Flash Plugin calls snd_async_add_pcm_handler() which always return -ENOSYS out of snd_pcm_ioplug_async(). This error causes Flash Plugin ALSA client code to fail. Is there a good reason why snd_pcm_ioplug_async() unconditionally return -ENOSYS? Is it possible to implement IO Plug async interface? Thank you for your help, Alexander Indenbaum

2 1

[alsa-devel] How to choose output channels for AD1386A sound?
by Ihar Hrachyshka 21 Apr '08

21 Apr '08

Hello! I build my BF533-EZKIT-based board (Blackfin processor development board) sound system on AD1386A sound. I use blackfin.uclinux.org distribution for it. One of my board's function is to play three different stereo sound streams simultaneously on it. What I done is choosing for AD1386 '3 stereos' option. After loading the kernel, my software is able to open pcm 3 times and plays sound properly, with no glitches or smth. The only missed functionality is that I can't specify the needed subdevice I want to play a stream on. I should have the possibility to play, f.e., mplayer stream on the 3rd stereo output while the 1st and the 2nd are not used. How can I do it? Also that would be great if I can optionally play the same stream on all the outputs. If I'm not wrong, this can be made through asound.conf settings though I can't figer out what are the right values. If I can choose next used output through kernel interface (kobject value or smth) then please tell me where should I look into. Here are some info (line-in playing on the 1st subdevice): root:~> aplay -l **** List of PLAYBACK Hardware Devices **** card 0: ad1836 [ADI ad1836], device 0: AD1836_PCM [AD1836_PCM] Subdevices: 2/3 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 root:~> cat /proc/asound/devices 0: [ 0] : control 16: [ 0- 0]: digital audio playback 24: [ 0- 0]: digital audio capture 33: : timer root:~> cat /proc/asound/pcm 00-00: AD1836_PCM : AD1836_PCM : playback 3 : capture 1 root:~> cat /proc/asound/cards 0 [ad1836 ]: AD1836 - ADI ad1836 ADI ad1836 at PF4 SPORT0 rx/tx dma 1/2 err irq 10 root:~> cat /proc/asound/ad1836/registers DAC_CTRL_1 0x0000 DAC_CTRL_2 0x1000 DAC_VOL_1L 0x23ff DAC_VOL_1R 0x33ff DAC_VOL_2L 0x43fa DAC_VOL_2R 0x53fa DAC_VOL_3L 0x63fa DAC_VOL_3R 0x73fa ADC_PEAK_1L 0xffff -63 dBFS ADC_PEAK_1R 0xffff -63 dBFS ADC_PEAK_2L 0xffff -63 dBFS ADC_PEAK_2R 0xffff -63 dBFS ADC_CTRL_1 0xc000 ADC_CTRL_2 0xd180 ADC_CTRL_3 0xe03a root:~> cat /proc/asound/ad1836/pcm0p/info card: 0 device: 0 subdevice: 0 stream: PLAYBACK id: AD1836_PCM name: AD1836_PCM subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 3 subdevices_avail: 2 root:~> ls /proc/asound/ad1836/pcm0p/ info oss sub0 sub1 sub2 xrun_debug root:~> cat /proc/asound/ad1836/pcm0p/sub0/status state: RUNNING trigger_time: 1208611073.963748000 tstamp : 1208611991.506408000 delay : 6441 avail : 1751 avail_max : 6188 ----- hw_ptr : 44035799 appl_ptr : 44042240 root:~> cat /proc/asound/ad1836/pcm0p/sub1/status closed root:~> cat /proc/asound/ad1836/pcm0p/sub2/status closed Thanks for help.

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[alsa-devel] [PATCH] speaker-test.c - fix sine generator on big-endian archs
by Giuliano Pochini 21 Apr '08

21 Apr '08

speaker-test doesn't work well when you choose a little-endian format on a big-endian processor, or the opposite. Yes, I know about plughw:, but for debugging purposes it may not be an option. The following patch add proper support for S32 and S16 support for but LE and BE processors: - The "if (bits-per-sample)" construct was replaced by case (format). - Support for S16_BE, S32_BE formats was added. - S16_LE and S32_LE were made compatible with big-endian processors. - NB: The pink noise generator wasn't changed (I'll do if this patch is OK). --- speaker-test/speaker-test.c__orig 2008-04-19 23:01:22.000000000 +0200 +++ speaker-test/speaker-test.c 2008-04-19 23:29:45.000000000 +0200 @@ -139,7 +139,6 @@ static void generate_sine(uint8_t *frame int16_t *samp16 = (int16_t*) frames; int32_t *samp32 = (int32_t*) frames; float *samp_f = (float*) frames; - int sample_size_bits = snd_pcm_format_width(format); while (count-- > 0) { //res = sin((phase * 2 * M_PI) / max_phase - M_PI) * 32767; @@ -154,26 +153,43 @@ static void generate_sine(uint8_t *frame //ires = 0; for(chn=0;chn<channels;chn++) { - if (sample_size_bits == 8) { + switch (format) { + case SND_PCM_FORMAT_S8: if (chn==channel) { res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * 0x03fffffff; /* Don't use MAX volume */ ires = res; - *samp8++ = ires >> 24; - //*samp8++ = 0x12; + *samp8++ = ires >> 24; } else { - *samp8++ = 0; + *samp8++ = 0; } - } else if (sample_size_bits == 16) { + break; + case SND_PCM_FORMAT_S16_LE: if (chn==channel) { res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * 0x03fffffff; /* Don't use MAX volume */ ires = res; - *samp16++ = ires >>16; - //*samp16++ = 0x1234; +#if __BYTE_ORDER == __LITTLE_ENDIAN + *samp16++ = ires >> 16; +#elif __BYTE_ORDER == __BIG_ENDIAN + *samp16++ = bswap_16(ires >> 16); +#endif } else { - //*samp16++ = (ires >>16)+1; - *samp16++ = 0; + *samp16++ = 0; } - } else if ((sample_size_bits == 32) && (format == SND_PCM_FORMAT_FLOAT_LE)) { + break; + case SND_PCM_FORMAT_S16_BE: + if (chn==channel) { + res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * 0x03fffffff; /* Don't use MAX volume */ + ires = res; +#if __BYTE_ORDER == __BIG_ENDIAN + *samp16++ = ires >> 16; +#elif __BYTE_ORDER == __LITTLE_ENDIAN + *samp16++ = bswap_16(ires >> 16); +#endif + } else { + *samp16++ = 0; + } + break; + case SND_PCM_FORMAT_FLOAT_LE: if (chn==channel) { res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * 0.75 ; /* Don't use MAX volume */ fres = res; @@ -185,18 +201,35 @@ static void generate_sine(uint8_t *frame //*samp32++ = ires; *samp_f++ = 0.0; } - } else if ((sample_size_bits == 32) && (format != SND_PCM_FORMAT_FLOAT_LE)) { + break; + case SND_PCM_FORMAT_S32_LE: if (chn==channel) { res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * 0x03fffffff; /* Don't use MAX volume */ ires = res; - *samp32++ = ires; - //*samp32++ = 0xF2345678; - //printf("res=%lf, ires=%d 0x%x, samp32=0x%x\n",res,ires, ires, samp32[-1]); +#if __BYTE_ORDER == __LITTLE_ENDIAN + *samp32++ = ires; +#elif __BYTE_ORDER == __BIG_ENDIAN + *samp32++ = bswap_32(ires); +#endif } else { - //*samp32++ = ires+0x10000; - //*samp32++ = ires; - *samp32++ = 0; + *samp32++ = 0; } + break; + case SND_PCM_FORMAT_S32_BE: + if (chn==channel) { + res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * 0x03fffffff; /* Don't use MAX volume */ + ires = res; +#if __BYTE_ORDER == __BIG_ENDIAN + *samp32++ = ires; +#elif __BYTE_ORDER == __LITTLE_ENDIAN + *samp32++ = bswap_32(ires); +#endif + } else { + *samp32++ = 0; + } + break; + default: + ; } } -- Giuliano.

2 1

[alsa-devel] Monotonic timestamps
by Lennart Poettering 21 Apr '08

21 Apr '08

Hey! I saw that ALSA 1.0.16 claims to support monotonic timestamps. I couldn't figure out however, how they are supposed to work. From the sources I only could figure out that they are enabled if a) the libc knows CLOCK_MONOTONIC a) the kernel supports CLOCK_MONOTONIC, b) the ALSA kernel code support timestamps this way. That's at least how I read this code from pcm_hw.c: <snip> #if defined(HAVE_CLOCK_GETTIME) && defined(CLOCK_MONOTONIC) if (SNDRV_PROTOCOL_VERSION(2, 0, 9) <= ver) { struct timespec timespec; if (clock_gettime(CLOCK_MONOTONIC, &timespec) == 0) { int on = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC; if (ioctl(fd, SNDRV_PCM_IOCTL_TTSTAMP, &on) < 0) { ret = -errno; SNDMSG("TTSTAMP failed\n"); return ret; } monotonic = 1; } } #endif else if (SNDRV_PROTOCOL_VERSION(2, 0, 5) <= ver) { int on = 1; if (ioctl(fd, SNDRV_PCM_IOCTL_TSTAMP, &on) < 0) { ret = -errno; SNDMSG("TSTAMP failed\n"); return ret; } } </snip> The problem with this is that there seems to be no way to determine from an application if monotonic timestamps are enabled in an snd_pcm_t or not, ALSA just switches over to them, making the timestamps relatively useless, because we cannot reliable relate them to timestamps we query from the kernel ourselves -- because we just don't know if we need to use CLOCK_MONOTONIC or CLOCK_REALTIME. I find it very strange that ALSA just switches to monotonic timestamps just like that, anyway. Programs written for wallclack timestamps will break if they run on a system where ALSA uses monotonic timestamps! There's a function missing that enables monotonic timestamps explicitly, or at least one that can be used to query if they are monotonic or not. Could anybody please explain the difference between status->tstamp and status->trigger_tstamp for me, please? The doxygen docs are bit too terse on this, I fear. Oh, and the code I pasted above will not compile if the #ifdef check fails, because of the "else". The "else" should be moved inside of #ifddef block. 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] snd_hda_intel locking up on VIA CX700M2 with VT1708A codec
by Simon Farnsworth 21 Apr '08

21 Apr '08

Hello, I'm trying to work out what's going wrong with the HDA controller in our VIA CX700M2 board; there's a full bug report at https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3910, but briefly, I'm seeing it lock up and refuse to DMA after long enough transferring audio data. Does anyone have any ideas that would help me track down this lockup and find a workaround? -- Thanks, Simon Farnsworth

1 2

[alsa-devel] [PATCH] asoc tlv320aic33: skip usage of PLL in some cases
by Daniel Mack 19 Apr '08

19 Apr '08

This patch makes the tlv320aic33 driver skip the initialisation of the PLL in case the sysclk is 256 * samplerate. Had to do some minor refactoring too - the check whether a sysclk set by set_sysclk() is valid is now done in set_hw_params(). Signed-off-by: Daniel Mack <daniel(a)caiaq.de>

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[alsa-devel] Patchmix-style application for EMU cards
by Camilo Polymeris 18 Apr '08

18 Apr '08

Hello I am working on a graphical mixing/routing application for the Emu10k series of semi-pro soundcards, but I am having trouble figuring out the routing mechanism of the driver, specially the "Multichannel PCM" part. Is there any technical documentation on this card I could read (I haven't found any), or maybe the driver developer can help me out? Thanks, Camilo.

2 2

[alsa-devel] [PATCH] asoc tlv320aic3x: add GPIO support
by Daniel Mack 18 Apr '08

18 Apr '08

This patch adds support for AIC3x GPIO lines. They can be configured for many possible functions as well as be driven manually. I also introduced i2c read functionality since the GPIO state register has to be read from hardware every time and can not be served from cache. Signed-off-by: Daniel Mack <daniel(a)caiaq.de>

1 0

[alsa-devel] Alsa Driver on Coldfire
by Simon Dunn 18 Apr '08

18 Apr '08

Hi I've been given the job of writing a sound card driver using the alsa architecture for a development platform that uses a coldfire chip. The first thing I came across was a chunky 'if !M68K' in sound/Kconfig, which excludes all the alsa components. I can't find any reason for this in the mailing list archives or on the web in general, so I'm wondering if anybody knows why the M68K architecture is excluded? Cheers Simon -- Simon Dunn, Software Engineer MPC Data Limited e-mail: sdunn(a)mpc-data.co.uk web: www.mpc-data.co.uk tel: +44 (0) 1225 710600 fax: +44 (0) 1225 710601 ddi: +44 (0) 1225 710631 MPC Data Limited is a company registered in England and Wales with company number 05507446 Registered Address: County Gate, County Way, Trowbridge, Wiltshire, BA14 7FJ VAT no: 850625238 The information in this email and in the attached documents is confidential and may be legally privileged. Any unauthorized review, copying, disclosure or distribution is prohibited and may be unlawful. It is intended solely for the addressee. Access to this email by anyone else is unauthorized. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. When addressed to our clients any opinions or advice contained in this email is subject to the terms and conditions expressed in the governing contract.

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[alsa-devel] [PATCH 001c/001] snd-powermac: more older kernel compatibility
by Risto Suominen 18 Apr '08

18 Apr '08

2008/4/17, Takashi Iwai <tiwai(a)suse.de>: > > The hack isn't definitely for the latest upstream, so if this is > needed, rather define in each *.c in alsa-driver/ppc please. > Now it's done in ppc/*.patch. Risto

2 1

[alsa-devel] [PATCH] SOC: fix tests in cs4270_hw_params()
by Roel Kluin 17 Apr '08

17 Apr '08

cs4270_hw_params does several times: ret = snd_soc_write() if (ret < 0) ... This only works when ret is signed. Signed-off-by: Roel Kluin <12o3l(a)tiscali.nl> --- diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index bf2ab72..e73fcfd 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -372,7 +372,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; struct cs4270_private *cs4270 = codec->private_data; - unsigned int ret = 0; + int ret; unsigned int i; unsigned int rate; unsigned int ratio;

3 2

Re: [alsa-devel] alsa 1.0.16 on kernel 2.6.23.9
by Takashi Iwai 17 Apr '08

17 Apr '08

At Thu, 17 Apr 2008 12:42:16 +0000, Vitaly V. Ch wrote: > > On Thu, Apr 17, 2008 at 12:13 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Thu, 17 Apr 2008 12:06:12 +0000, > > > > Vitaly V. Ch wrote: > > > > > > On Thu, Apr 17, 2008 at 11:54 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > At Thu, 17 Apr 2008 11:14:36 +0000, > > > > > > > > Vitaly V. Ch wrote: > > > > > > > > > > On Thu, Apr 17, 2008 at 9:55 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > > > At Thu, 17 Apr 2008 07:26:25 +0000, > > > > > > > > > > > > Vitaly V. Ch wrote: > > > > > > > > > > > > > > Have anyone successfully run alsa 1.0.16 on kernel 2.6.23.9? > > > > > > > > > > > > > > after success configure and make with one warning: > > > > > > > > > > > > > > WARNING: /tmp/alsa-driver-1.0.16/misc/ac97_bus: 'ac97_bus_type' > > > > > > > exported twice. Previous export was in vmlinux > > > > > > > > > > > > Do you have CONFIG_AC97_BUS=y? This should be a module, too. > > > > > Yes, I have it. I can't find in menuconfig item to switch it. Can I > > > > > switch it manually by editing .config? > > > > > > > > Yes. But usually it's selected by other items. So you likely made > > > > dependent components all built-in (=y). They should be also modules. > > > where I can see which components require to be modules also? > > > > Read Kconfig files. > I grep all Kconfig files and found out nothing depended on CONFIG_AC97_BUS Others do "select" CONFIG_AC97_BUS. > > > I don't compile OSS at all and all alsa is modules. which component > > > can be depend on it? > > > > Don't know unless you expose your config. Just try to select m and > > rebuild. > .config is attached. > I try to select m and make - make revert m to y before begin compile. > > may be I must turn off support of sound in kernel at all? No, it won't work, at least the sound core must be enabled. Takashi

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[alsa-devel] alsa 1.0.16 on kernel 2.6.23.9
by Vitaly V. Ch 17 Apr '08

17 Apr '08

Have anyone successfully run alsa 1.0.16 on kernel 2.6.23.9? after success configure and make with one warning: WARNING: /tmp/alsa-driver-1.0.16/misc/ac97_bus: 'ac97_bus_type' exported twice. Previous export was in vmlinux I receive non-working installation of alsa Any ideas? \\wbr Vitaly Chernookiy

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[alsa-devel] External PCM IO Plugin development
by Alexander Indenbaum 17 Apr '08

17 Apr '08

Takashi & alsa-devel, I am playing with embedded Linux box based on kernel 2.4.36. Our box is equipped with proprietary chip which allows PCM playback using user space DMA pool API. My target is to enable audio playback for ALSA clients by redirecting PCM stream into proprietary chip. I did a little RTFM and with a lot of help from this list discovered that fortunately ALSA expendable architecture allows such things as a part of PCM External Plugin SDK. So I downloaded and installed following ALSA 1.0.16 distribution components: - driver configured with "./configure --with-cards=seq-dummy,dummy --with-kernel=/path/to/my/linux-2.4.36" - lib configured with "./configure --with-pcm-plugins=all" I looked at Jack IO plugin at plugins and started with skeleton plugin of my own: pcm_sigma.c (attached). I hope to redirect PCM into my plugin using following /etc/asound.conf ============= pcm.!default { type sigma } ============= What I see when I fire ALSA client ( friefox with YouTube flash ) that ALSA 1. loads/links my module: ALSA lib dlmisc.c:51:(snd_dlopen) snd_dlopen name = (/usr/lib/alsa-lib/libasound_module_pcm_sigma.so) 2. calls open interface of the module: _snd_pcm_sigma_open @ pcm_sigma.c:118 snd_pcm_sigma_open @ pcm_sigma.c:86 3. calls close interface of the module: snd_pcm_sigma_close @ pcm_sigma.c:23 4. goto 2 So basically what I see is an infinite open/close loop without any call to other interfaces of the module. I hoped to get my hands on PCM stream and start to push it into our proprietary chip :(. QUESTIONS: Is my PCM IO Plugin approach correct? Is my configuration (/etc/asound.conf) correct? Is my implemention of PCM IO plugin interface correct? Should I implement additional methods? What do you think causes infinite open/close loop? Thank you for your help, Alexander Indenbaum

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[alsa-devel] [PATCH 001b/001] snd-powermac: older kernel compatibility
by Risto Suominen 17 Apr '08

17 Apr '08

I had to do these small modifications to make it work with older kernels. Risto

2 3

[alsa-devel] [PATCH 002a/003] snd-powermac: AWACS and Screamer mixers for PM7500, Beige, and iMac SL
by Risto Suominen 16 Apr '08

16 Apr '08

From: Risto Suominen <Risto.Suominen(a)gmail.com> Add mixer controls and correct headphone detection bits for PowerMacs 7300/7500 (AWACS) and G3 Beige (Screamer), and iMac G3 Slot-loading (Screamer). Signed-off-by: Risto Suominen <Risto.Suominen(a)gmail.com> --- Kernel version 2.6.25-rc8 (probably any 2.6) --- sound/ppc/awacs.c.orig 2008-04-05 17:45:08.000000000 +0300 +++ sound/ppc/awacs.c 2008-04-07 11:24:08.108621048 +0300 @@ -493,14 +500,14 @@ static struct snd_kcontrol_new snd_pmac_ .private_value = AMP_CH_HD, }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Tone Control - Bass", + .name = "Tone Control - Bass Playback Volume", .info = snd_pmac_awacs_info_tone_amp, .get = snd_pmac_awacs_get_tone_amp, .put = snd_pmac_awacs_put_tone_amp, .private_value = 0, }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Tone Control - Treble", + .name = "Tone Control - Treble Playback Volume", .info = snd_pmac_awacs_info_tone_amp, .get = snd_pmac_awacs_get_tone_amp, .put = snd_pmac_awacs_put_tone_amp, @@ -544,7 +551,7 @@ static int snd_pmac_screamer_mic_boost_i uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; uinfo->value.integer.min = 0; - uinfo->value.integer.max = 2; + uinfo->value.integer.max = 3; return 0; } @@ -552,16 +559,14 @@ static int snd_pmac_screamer_mic_boost_g struct snd_ctl_elem_value *ucontrol) { struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); - int val; + int val = 0; unsigned long flags; spin_lock_irqsave(&chip->reg_lock, flags); if (chip->awacs_reg[6] & MASK_MIC_BOOST) - val = 2; - else if (chip->awacs_reg[0] & MASK_GAINLINE) - val = 1; - else - val = 0; + val |= 2; + if (chip->awacs_reg[0] & MASK_GAINLINE) + val |= 1; spin_unlock_irqrestore(&chip->reg_lock, flags); ucontrol->value.integer.value[0] = val; return 0; @@ -578,11 +583,10 @@ static int snd_pmac_screamer_mic_boost_p spin_lock_irqsave(&chip->reg_lock, flags); val0 = chip->awacs_reg[0] & ~MASK_GAINLINE; val6 = chip->awacs_reg[6] & ~MASK_MIC_BOOST; - if (ucontrol->value.integer.value[0] > 0) { + if (ucontrol->value.integer.value[0] & 1) val0 |= MASK_GAINLINE; - if (ucontrol->value.integer.value[0] > 1) - val6 |= MASK_MIC_BOOST; - } + if (ucontrol->value.integer.value[0] & 2) + val6 |= MASK_MIC_BOOST; if (val0 != chip->awacs_reg[0]) { snd_pmac_awacs_write_reg(chip, 0, val0); changed = 1; @@ -599,9 +603,32 @@ static int snd_pmac_screamer_mic_boost_p * lists of mixer elements */ static struct snd_kcontrol_new snd_pmac_awacs_mixers[] __initdata = { - AWACS_VOLUME("Master Playback Volume", 2, 6, 1), AWACS_SWITCH("Master Capture Switch", 1, SHIFT_LOOPTHRU, 0), - AWACS_VOLUME("Capture Volume", 0, 4, 0), + AWACS_VOLUME("Master Capture Volume", 0, 4, 0), +/* AWACS_SWITCH("Unknown Playback Switch", 6, SHIFT_PAROUT0, 0), */ +}; + +static struct snd_kcontrol_new snd_pmac_screamer_mixers_beige[] __initdata = { + AWACS_VOLUME("Master Playback Volume", 2, 6, 1), + AWACS_VOLUME("Play-through Playback Volume", 5, 6, 1), + AWACS_SWITCH("Line Capture Switch", 0, SHIFT_MUX_MIC, 0), + AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_LINE, 0), +}; + +static struct snd_kcontrol_new snd_pmac_screamer_mixers_imac[] __initdata = { + AWACS_VOLUME("Line out Playback Volume", 2, 6, 1), + AWACS_VOLUME("Master Playback Volume", 5, 6, 1), + AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0), +}; + +static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac7500[] __initdata = { + AWACS_VOLUME("Line out Playback Volume", 2, 6, 1), + AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0), + AWACS_SWITCH("Line Capture Switch", 0, SHIFT_MUX_MIC, 0), +}; + +static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac[] __initdata = { + AWACS_VOLUME("Master Playback Volume", 2, 6, 1), AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0), }; @@ -621,25 +648,49 @@ static struct snd_kcontrol_new snd_pmac_ static struct snd_kcontrol_new snd_pmac_awacs_master_sw __initdata = AWACS_SWITCH("Master Playback Switch", 1, SHIFT_HDMUTE, 1); +static struct snd_kcontrol_new snd_pmac_awacs_master_sw_imac __initdata = +AWACS_SWITCH("Line out Playback Switch", 1, SHIFT_HDMUTE, 1); + static struct snd_kcontrol_new snd_pmac_awacs_mic_boost[] __initdata = { - AWACS_SWITCH("Mic Boost", 0, SHIFT_GAINLINE, 0), + AWACS_SWITCH("Mic Boost Capture Switch", 0, SHIFT_GAINLINE, 0), }; static struct snd_kcontrol_new snd_pmac_screamer_mic_boost[] __initdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Mic Boost", + .name = "Mic Boost Capture Volume", .info = snd_pmac_screamer_mic_boost_info, .get = snd_pmac_screamer_mic_boost_get, .put = snd_pmac_screamer_mic_boost_put, }, }; +static struct snd_kcontrol_new snd_pmac_awacs_mic_boost_pmac7500[] __initdata = +{ + AWACS_SWITCH("Line Boost Capture Switch", 0, SHIFT_GAINLINE, 0), +}; + +static struct snd_kcontrol_new snd_pmac_screamer_mic_boost_beige[] __initdata = +{ + AWACS_SWITCH("Line Boost Capture Switch", 0, SHIFT_GAINLINE, 0), + AWACS_SWITCH("CD Boost Capture Switch", 6, SHIFT_MIC_BOOST, 0), +}; + +static struct snd_kcontrol_new snd_pmac_screamer_mic_boost_imac[] __initdata = +{ + AWACS_SWITCH("Line Boost Capture Switch", 0, SHIFT_GAINLINE, 0), + AWACS_SWITCH("Mic Boost Capture Switch", 6, SHIFT_MIC_BOOST, 0), +}; + static struct snd_kcontrol_new snd_pmac_awacs_speaker_vol[] __initdata = { AWACS_VOLUME("PC Speaker Playback Volume", 4, 6, 1), }; + static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw __initdata = AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_SPKMUTE, 1); +static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw_imac __initdata = +AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_PAROUT1, 0); + /* * add new mixer elements to the card @@ -708,6 +763,14 @@ static void snd_pmac_awacs_resume(struct } #endif /* CONFIG_PM */ +#define IS_PM7500 (machine_is_compatible("AAPL,7500")) +#define IS_BEIGE (machine_is_compatible("AAPL,Gossamer")) +#define IS_IMAC (machine_is_compatible("PowerMac2,1") \ + || machine_is_compatible("PowerMac2,2") \ + || machine_is_compatible("PowerMac4,1")) + +static int imac; + #ifdef PMAC_SUPPORT_AUTOMUTE /* * auto-mute stuffs @@ -750,9 +813,16 @@ static void snd_pmac_awacs_update_automu } else #endif { - int reg = chip->awacs_reg[1] | (MASK_HDMUTE|MASK_SPKMUTE); + int reg = chip->awacs_reg[1] + | (MASK_HDMUTE | MASK_SPKMUTE); + if (imac) { + reg &= ~MASK_SPKMUTE; + reg &= ~MASK_PAROUT1; + } if (snd_pmac_awacs_detect_headphone(chip)) reg &= ~MASK_HDMUTE; + else if (imac) + reg |= MASK_PAROUT1; else reg &= ~MASK_SPKMUTE; if (do_notify && reg == chip->awacs_reg[1]) @@ -778,8 +848,11 @@ static void snd_pmac_awacs_update_automu int __init snd_pmac_awacs_init(struct snd_pmac *chip) { + int pm7500 = IS_PM7500; + int beige = IS_BEIGE; int err, vol; + imac = IS_IMAC; /* looks like MASK_GAINLINE triggers something, so we set here * as start-up */ @@ -826,20 +902,25 @@ snd_pmac_awacs_init(struct snd_pmac *chi /* set headphone-jack detection bit */ switch (chip->model) { case PMAC_AWACS: - chip->hp_stat_mask = 0x04; + chip->hp_stat_mask = pm7500 ? MASK_HDPCONN + : MASK_LOCONN; break; case PMAC_SCREAMER: switch (chip->device_id) { case 0x08: - /* 1 = side jack, 2 = front jack */ - chip->hp_stat_mask = 0x03; + case 0x0B: + chip->hp_stat_mask = imac + ? MASK_LOCONN_IMAC | + MASK_HDPLCONN_IMAC | + MASK_HDPRCONN_IMAC + : MASK_HDPCONN; break; case 0x00: case 0x05: - chip->hp_stat_mask = 0x04; + chip->hp_stat_mask = MASK_LOCONN; break; default: - chip->hp_stat_mask = 0x08; + chip->hp_stat_mask = MASK_HDPCONN; break; } break; @@ -857,15 +938,37 @@ snd_pmac_awacs_init(struct snd_pmac *chi if ((err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_mixers), snd_pmac_awacs_mixers)) < 0) return err; - if (chip->model == PMAC_SCREAMER) + if (beige) + ; + else if (chip->model == PMAC_SCREAMER) err = build_mixers(chip, ARRAY_SIZE(snd_pmac_screamer_mixers2), snd_pmac_screamer_mixers2); - else + else if (!pm7500) err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_mixers2), snd_pmac_awacs_mixers2); if (err < 0) return err; - chip->master_sw_ctl = snd_ctl_new1(&snd_pmac_awacs_master_sw, chip); + if (pm7500) + err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_awacs_mixers_pmac7500), + snd_pmac_awacs_mixers_pmac7500); + else if (beige) + err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_screamer_mixers_beige), + snd_pmac_screamer_mixers_beige); + else if (imac) + err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_screamer_mixers_imac), + snd_pmac_screamer_mixers_imac); + else + err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_awacs_mixers_pmac), + snd_pmac_awacs_mixers_pmac); + if (err < 0) + return err; + chip->master_sw_ctl = snd_ctl_new1((pm7500 || imac) + ? &snd_pmac_awacs_master_sw_imac + : &snd_pmac_awacs_master_sw, chip); if ((err = snd_ctl_add(chip->card, chip->master_sw_ctl)) < 0) return err; #ifdef PMAC_AMP_AVAIL @@ -893,14 +1000,32 @@ snd_pmac_awacs_init(struct snd_pmac *chi if ((err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_speaker_vol), snd_pmac_awacs_speaker_vol)) < 0) return err; - chip->speaker_sw_ctl = snd_ctl_new1(&snd_pmac_awacs_speaker_sw, chip); + chip->speaker_sw_ctl = snd_ctl_new1(imac + ? &snd_pmac_awacs_speaker_sw_imac + : &snd_pmac_awacs_speaker_sw, chip); if ((err = snd_ctl_add(chip->card, chip->speaker_sw_ctl)) < 0) return err; } - if (chip->model == PMAC_SCREAMER) { - if ((err = build_mixers(chip, ARRAY_SIZE(snd_pmac_screamer_mic_boost), - snd_pmac_screamer_mic_boost)) < 0) + if (beige) { + if ((err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_screamer_mic_boost_beige), + snd_pmac_screamer_mic_boost_beige)) < 0) + return err; + } else if (imac) { + if ((err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_screamer_mic_boost_imac), + snd_pmac_screamer_mic_boost_imac)) < 0) + return err; + } else if (chip->model == PMAC_SCREAMER) { + if ((err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_screamer_mic_boost), + snd_pmac_screamer_mic_boost)) < 0) + return err; + } else if (pm7500) { + if ((err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_awacs_mic_boost_pmac7500), + snd_pmac_awacs_mic_boost_pmac7500)) < 0) return err; } else { if ((err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_mic_boost), --- sound/ppc/awacs.h.orig 2008-04-05 17:45:08.000000000 +0300 +++ sound/ppc/awacs.h 2008-04-07 11:26:58.198763416 +0300 @@ -116,6 +116,11 @@ struct awacs_regs { #define MASK_HDMUTE MASK_AMUTE #define SHIFT_HDMUTE 9 #define MASK_PAROUT (0x3 << 10) /* Parallel Out (???) */ +#define MASK_PAROUT0 (0x1 << 10) /* Parallel Out (???) */ +#define MASK_PAROUT1 (0x1 << 11) /* Parallel Out (enable speaker) */ +#define SHIFT_PAROUT 10 +#define SHIFT_PAROUT0 10 +#define SHIFT_PAROUT1 11 #define SAMPLERATE_48000 (0x0 << 3) /* 48 or 44.1 kHz */ #define SAMPLERATE_32000 (0x1 << 3) /* 32 or 29.4 kHz */ @@ -152,8 +157,15 @@ struct awacs_regs { #define MASK_REVISION (0xf << 12) /* Revision Number */ #define MASK_MFGID (0xf << 8) /* Mfg. ID */ #define MASK_CODSTATRES (0xf << 4) /* bits 4 - 7 reserved */ -#define MASK_INPPORT (0xf) /* Input Port */ -#define MASK_HDPCONN 8 /* headphone plugged in */ +#define MASK_INSENSE (0xf) /* port sense bits: */ +#define MASK_HDPCONN 8 /* headphone plugged in */ +#define MASK_LOCONN 4 /* line-out plugged in */ +#define MASK_LICONN 2 /* line-in plugged in */ +#define MASK_MICCONN 1 /* microphone plugged in */ +#define MASK_LICONN_IMAC 8 /* line-in plugged in */ +#define MASK_HDPRCONN_IMAC 4 /* headphone right plugged in */ +#define MASK_HDPLCONN_IMAC 2 /* headphone left plugged in */ +#define MASK_LOCONN_IMAC 1 /* line-out plugged in */ /* Clipping Count Reg Bit Masks */ /* -------- ----- --- --- ----- */

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[alsa-devel] [PATCH 003b/003] snd-powermac: style burgundy.c
by Risto Suominen 16 Apr '08

16 Apr '08

From: Risto Suominen <Risto.Suominen(a)gmail.com> Coding style corrections for burgundy.c. Signed-off-by: Risto Suominen <Risto.Suominen(a)gmail.com> --- Kernel version 2.6.25-rc8 (probably any 2.6) --- sound/ppc/burgundy.c.orig 2008-04-05 17:45:08.000000000 +0300 +++ sound/ppc/burgundy.c 2008-04-07 12:00:20.893307568 +0300 @@ -102,7 +102,8 @@ snd_pmac_burgundy_rcw(struct snd_pmac *c } static void -snd_pmac_burgundy_wcb(struct snd_pmac *chip, unsigned int addr, unsigned int val) +snd_pmac_burgundy_wcb(struct snd_pmac *chip, unsigned int addr, + unsigned int val) { out_le32(&chip->awacs->codec_ctrl, addr + 0x300000 + (val & 0xff)); snd_pmac_burgundy_busy_wait(chip); @@ -126,8 +127,11 @@ snd_pmac_burgundy_rcb(struct snd_pmac *c return val; } +#define BASE2ADDR(base) ((base) << 12) +#define ADDR2BASE(addr) ((addr) >> 12) + /* - * Burgundy volume: 0 - 100, stereo + * Burgundy volume: 0 - 100, stereo, word reg */ static void snd_pmac_burgundy_write_volume(struct snd_pmac *chip, unsigned int address, @@ -168,13 +172,6 @@ snd_pmac_burgundy_read_volume(struct snd volume[1] = 0; } - -/* - */ - -#define BASE2ADDR(base) ((base) << 12) -#define ADDR2BASE(addr) ((addr) >> 12) - static int snd_pmac_burgundy_info_volume(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -191,8 +188,8 @@ static int snd_pmac_burgundy_get_volume( struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); unsigned int addr = BASE2ADDR(kcontrol->private_value & 0xff); int shift = (kcontrol->private_value >> 8) & 0xff; - snd_pmac_burgundy_read_volume(chip, addr, ucontrol->value.integer.value, - shift); + snd_pmac_burgundy_read_volume(chip, addr, + ucontrol->value.integer.value, shift); return 0; } @@ -204,8 +201,8 @@ static int snd_pmac_burgundy_put_volume( int shift = (kcontrol->private_value >> 8) & 0xff; long nvoices[2]; - snd_pmac_burgundy_write_volume(chip, addr, ucontrol->value.integer.value, - shift); + snd_pmac_burgundy_write_volume(chip, addr, + ucontrol->value.integer.value, shift); snd_pmac_burgundy_read_volume(chip, addr, nvoices, shift); return (nvoices[0] != ucontrol->value.integer.value[0] || nvoices[1] != ucontrol->value.integer.value[1]);

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[alsa-devel] [PATCH 001a/001] snd-powermac: enable headphone detection on older kernels
by Risto Suominen 16 Apr '08

16 Apr '08

So this is not for kernel tree, but alsa-driver. Risto

1 0

[alsa-devel] [PATCH 003a/003] snd-powermac: Burgundy mixers for B&W and iMac
by Risto Suominen 16 Apr '08

16 Apr '08

From: Risto Suominen <Risto.Suominen(a)gmail.com> Add mixer controls and correct headphone detection bits for PowerMac G3 B&W and iMac G3 Tray-loading, both having Burgundy chipset. Signed-off-by: Risto Suominen <Risto.Suominen(a)gmail.com> --- Kernel version 2.6.25-rc8 (probably any 2.6) --- sound/ppc/burgundy.c.orig 2008-04-05 17:45:08.000000000 +0300 +++ sound/ppc/burgundy.c 2008-04-07 12:00:20.893307568 +0300 @@ -211,17 +208,156 @@ static int snd_pmac_burgundy_put_volume( nvoices[1] != ucontrol->value.integer.value[1]); } -#define BURGUNDY_VOLUME(xname, xindex, addr, shift) \ +#define BURGUNDY_VOLUME_W(xname, xindex, addr, shift) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex,\ .info = snd_pmac_burgundy_info_volume,\ .get = snd_pmac_burgundy_get_volume,\ .put = snd_pmac_burgundy_put_volume,\ .private_value = ((ADDR2BASE(addr) & 0xff) | ((shift) << 8)) } -/* lineout/speaker */ +/* + * Burgundy volume: 0 - 100, stereo, 2-byte reg + */ +static void +snd_pmac_burgundy_write_volume_2b(struct snd_pmac *chip, unsigned int address, + long *volume, int off) +{ + int lvolume, rvolume; + + off |= off << 2; + lvolume = volume[0] ? volume[0] + BURGUNDY_VOLUME_OFFSET : 0; + rvolume = volume[1] ? volume[1] + BURGUNDY_VOLUME_OFFSET : 0; + + snd_pmac_burgundy_wcb(chip, address + off, lvolume); + snd_pmac_burgundy_wcb(chip, address + off + 0x500, rvolume); +} + +static void +snd_pmac_burgundy_read_volume_2b(struct snd_pmac *chip, unsigned int address, + long *volume, int off) +{ + volume[0] = snd_pmac_burgundy_rcb(chip, address + off); + if (volume[0] >= BURGUNDY_VOLUME_OFFSET) + volume[0] -= BURGUNDY_VOLUME_OFFSET; + else + volume[0] = 0; + volume[1] = snd_pmac_burgundy_rcb(chip, address + off + 0x100); + if (volume[1] >= BURGUNDY_VOLUME_OFFSET) + volume[1] -= BURGUNDY_VOLUME_OFFSET; + else + volume[1] = 0; +} + +static int snd_pmac_burgundy_info_volume_2b(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_pmac_burgundy_get_volume_2b(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); + unsigned int addr = BASE2ADDR(kcontrol->private_value & 0xff); + int off = kcontrol->private_value & 0x300; + snd_pmac_burgundy_read_volume_2b(chip, addr, + ucontrol->value.integer.value, off); + return 0; +} + +static int snd_pmac_burgundy_put_volume_2b(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); + unsigned int addr = BASE2ADDR(kcontrol->private_value & 0xff); + int off = kcontrol->private_value & 0x300; + long nvoices[2]; + + snd_pmac_burgundy_write_volume_2b(chip, addr, + ucontrol->value.integer.value, off); + snd_pmac_burgundy_read_volume_2b(chip, addr, nvoices, off); + return (nvoices[0] != ucontrol->value.integer.value[0] || + nvoices[1] != ucontrol->value.integer.value[1]); +} + +#define BURGUNDY_VOLUME_2B(xname, xindex, addr, off) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex,\ + .info = snd_pmac_burgundy_info_volume_2b,\ + .get = snd_pmac_burgundy_get_volume_2b,\ + .put = snd_pmac_burgundy_put_volume_2b,\ + .private_value = ((ADDR2BASE(addr) & 0xff) | ((off) << 8)) } + +/* + * Burgundy gain/attenuation: 0 - 15, mono/stereo, byte reg + */ +static int snd_pmac_burgundy_info_gain(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int stereo = (kcontrol->private_value >> 24) & 1; + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = stereo + 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 15; + return 0; +} + +static int snd_pmac_burgundy_get_gain(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); + unsigned int addr = BASE2ADDR(kcontrol->private_value & 0xff); + int stereo = (kcontrol->private_value >> 24) & 1; + int atten = (kcontrol->private_value >> 25) & 1; + int oval; + + oval = snd_pmac_burgundy_rcb(chip, addr); + if (atten) + oval = ~oval & 0xff; + ucontrol->value.integer.value[0] = oval & 0xf; + if (stereo) + ucontrol->value.integer.value[1] = (oval >> 4) & 0xf; + return 0; +} -static int snd_pmac_burgundy_info_switch_out(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) +static int snd_pmac_burgundy_put_gain(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); + unsigned int addr = BASE2ADDR(kcontrol->private_value & 0xff); + int stereo = (kcontrol->private_value >> 24) & 1; + int atten = (kcontrol->private_value >> 25) & 1; + int oval, val; + + oval = snd_pmac_burgundy_rcb(chip, addr); + if (atten) + oval = ~oval & 0xff; + val = ucontrol->value.integer.value[0]; + if (stereo) + val |= ucontrol->value.integer.value[1] << 4; + else + val |= ucontrol->value.integer.value[0] << 4; + if (atten) + val = ~val & 0xff; + snd_pmac_burgundy_wcb(chip, addr, val); + return val != oval; +} + +#define BURGUNDY_VOLUME_B(xname, xindex, addr, stereo, atten) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex,\ + .info = snd_pmac_burgundy_info_gain,\ + .get = snd_pmac_burgundy_get_gain,\ + .put = snd_pmac_burgundy_put_gain,\ + .private_value = (ADDR2BASE(addr) | ((stereo) << 24) | ((atten) << 25)) } + +/* + * Burgundy switch: 0/1, mono/stereo, word reg + */ +static int snd_pmac_burgundy_info_switch_w(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { int stereo = (kcontrol->private_value >> 24) & 1; uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; @@ -231,111 +367,207 @@ static int snd_pmac_burgundy_info_switch return 0; } -static int snd_pmac_burgundy_get_switch_out(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int snd_pmac_burgundy_get_switch_w(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); - int lmask = kcontrol->private_value & 0xff; - int rmask = (kcontrol->private_value >> 8) & 0xff; + unsigned int addr = BASE2ADDR((kcontrol->private_value >> 16) & 0xff); + int lmask = 1 << (kcontrol->private_value & 0xff); + int rmask = 1 << ((kcontrol->private_value >> 8) & 0xff); int stereo = (kcontrol->private_value >> 24) & 1; - int val = snd_pmac_burgundy_rcb(chip, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES); + int val = snd_pmac_burgundy_rcw(chip, addr); ucontrol->value.integer.value[0] = (val & lmask) ? 1 : 0; if (stereo) ucontrol->value.integer.value[1] = (val & rmask) ? 1 : 0; return 0; } -static int snd_pmac_burgundy_put_switch_out(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int snd_pmac_burgundy_put_switch_w(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); - int lmask = kcontrol->private_value & 0xff; - int rmask = (kcontrol->private_value >> 8) & 0xff; + unsigned int addr = BASE2ADDR((kcontrol->private_value >> 16) & 0xff); + int lmask = 1 << (kcontrol->private_value & 0xff); + int rmask = 1 << ((kcontrol->private_value >> 8) & 0xff); int stereo = (kcontrol->private_value >> 24) & 1; int val, oval; - oval = snd_pmac_burgundy_rcb(chip, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES); - val = oval & ~(lmask | rmask); + oval = snd_pmac_burgundy_rcw(chip, addr); + val = oval & ~(lmask | (stereo ? rmask : 0)); if (ucontrol->value.integer.value[0]) val |= lmask; if (stereo && ucontrol->value.integer.value[1]) val |= rmask; - snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, val); + snd_pmac_burgundy_wcw(chip, addr, val); return val != oval; } -#define BURGUNDY_OUTPUT_SWITCH(xname, xindex, lmask, rmask, stereo) \ +#define BURGUNDY_SWITCH_W(xname, xindex, addr, lbit, rbit, stereo) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex,\ - .info = snd_pmac_burgundy_info_switch_out,\ - .get = snd_pmac_burgundy_get_switch_out,\ - .put = snd_pmac_burgundy_put_switch_out,\ - .private_value = ((lmask) | ((rmask) << 8) | ((stereo) << 24)) } - -/* line/speaker output volume */ -static int snd_pmac_burgundy_info_volume_out(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) + .info = snd_pmac_burgundy_info_switch_w,\ + .get = snd_pmac_burgundy_get_switch_w,\ + .put = snd_pmac_burgundy_put_switch_w,\ + .private_value = ((lbit) | ((rbit) << 8)\ + | (ADDR2BASE(addr) << 16) | ((stereo) << 24)) } + +/* + * Burgundy switch: 0/1, mono/stereo, byte reg, bit mask + */ +static int snd_pmac_burgundy_info_switch_b(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { int stereo = (kcontrol->private_value >> 24) & 1; - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; uinfo->count = stereo + 1; uinfo->value.integer.min = 0; - uinfo->value.integer.max = 15; + uinfo->value.integer.max = 1; return 0; } -static int snd_pmac_burgundy_get_volume_out(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int snd_pmac_burgundy_get_switch_b(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); - unsigned int addr = BASE2ADDR(kcontrol->private_value & 0xff); + unsigned int addr = BASE2ADDR((kcontrol->private_value >> 16) & 0xff); + int lmask = kcontrol->private_value & 0xff; + int rmask = (kcontrol->private_value >> 8) & 0xff; int stereo = (kcontrol->private_value >> 24) & 1; - int oval; - - oval = ~snd_pmac_burgundy_rcb(chip, addr) & 0xff; - ucontrol->value.integer.value[0] = oval & 0xf; + int val = snd_pmac_burgundy_rcb(chip, addr); + ucontrol->value.integer.value[0] = (val & lmask) ? 1 : 0; if (stereo) - ucontrol->value.integer.value[1] = (oval >> 4) & 0xf; + ucontrol->value.integer.value[1] = (val & rmask) ? 1 : 0; return 0; } -static int snd_pmac_burgundy_put_volume_out(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int snd_pmac_burgundy_put_switch_b(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); - unsigned int addr = BASE2ADDR(kcontrol->private_value & 0xff); + unsigned int addr = BASE2ADDR((kcontrol->private_value >> 16) & 0xff); + int lmask = kcontrol->private_value & 0xff; + int rmask = (kcontrol->private_value >> 8) & 0xff; int stereo = (kcontrol->private_value >> 24) & 1; - unsigned int oval, val; - - oval = ~snd_pmac_burgundy_rcb(chip, addr) & 0xff; - val = ucontrol->value.integer.value[0] & 15; - if (stereo) - val |= (ucontrol->value.integer.value[1] & 15) << 4; - else - val |= val << 4; - val = ~val & 0xff; + int val, oval; + oval = snd_pmac_burgundy_rcb(chip, addr); + val = oval & ~(lmask | rmask); + if (ucontrol->value.integer.value[0]) + val |= lmask; + if (stereo && ucontrol->value.integer.value[1]) + val |= rmask; snd_pmac_burgundy_wcb(chip, addr, val); return val != oval; } -#define BURGUNDY_OUTPUT_VOLUME(xname, xindex, addr, stereo) \ +#define BURGUNDY_SWITCH_B(xname, xindex, addr, lmask, rmask, stereo) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex,\ - .info = snd_pmac_burgundy_info_volume_out,\ - .get = snd_pmac_burgundy_get_volume_out,\ - .put = snd_pmac_burgundy_put_volume_out,\ - .private_value = (ADDR2BASE(addr) | ((stereo) << 24)) } + .info = snd_pmac_burgundy_info_switch_b,\ + .get = snd_pmac_burgundy_get_switch_b,\ + .put = snd_pmac_burgundy_put_switch_b,\ + .private_value = ((lmask) | ((rmask) << 8)\ + | (ADDR2BASE(addr) << 16) | ((stereo) << 24)) } +/* + * Burgundy mixers + */ static struct snd_kcontrol_new snd_pmac_burgundy_mixers[] __initdata = { - BURGUNDY_VOLUME("Master Playback Volume", 0, MASK_ADDR_BURGUNDY_MASTER_VOLUME, 8), - BURGUNDY_VOLUME("Line Playback Volume", 0, MASK_ADDR_BURGUNDY_VOLLINE, 16), - BURGUNDY_VOLUME("CD Playback Volume", 0, MASK_ADDR_BURGUNDY_VOLCD, 16), - BURGUNDY_VOLUME("Mic Playback Volume", 0, MASK_ADDR_BURGUNDY_VOLMIC, 16), - BURGUNDY_OUTPUT_VOLUME("PC Speaker Playback Volume", 0, MASK_ADDR_BURGUNDY_ATTENHP, 0), - /*BURGUNDY_OUTPUT_VOLUME("PCM Playback Volume", 0, MASK_ADDR_BURGUNDY_ATTENLINEOUT, 1),*/ - BURGUNDY_OUTPUT_VOLUME("Headphone Playback Volume", 0, MASK_ADDR_BURGUNDY_ATTENSPEAKER, 1), -}; -static struct snd_kcontrol_new snd_pmac_burgundy_master_sw __initdata = -BURGUNDY_OUTPUT_SWITCH("Headphone Playback Switch", 0, BURGUNDY_OUTPUT_LEFT, BURGUNDY_OUTPUT_RIGHT, 1); -static struct snd_kcontrol_new snd_pmac_burgundy_speaker_sw __initdata = -BURGUNDY_OUTPUT_SWITCH("PC Speaker Playback Switch", 0, BURGUNDY_OUTPUT_INTERN, 0, 0); + BURGUNDY_VOLUME_W("Master Playback Volume", 0, + MASK_ADDR_BURGUNDY_MASTER_VOLUME, 8), + BURGUNDY_VOLUME_W("CD Capture Volume", 0, + MASK_ADDR_BURGUNDY_VOLCD, 16), + BURGUNDY_VOLUME_2B("Input Capture Volume", 0, + MASK_ADDR_BURGUNDY_VOLMIX01, 2), + BURGUNDY_VOLUME_2B("Mixer Playback Volume", 0, + MASK_ADDR_BURGUNDY_VOLMIX23, 0), + BURGUNDY_VOLUME_B("CD Gain Capture Volume", 0, + MASK_ADDR_BURGUNDY_GAINCD, 1, 0), + BURGUNDY_SWITCH_W("Master Capture Switch", 0, + MASK_ADDR_BURGUNDY_OUTPUTENABLES, 24, 0, 0), + BURGUNDY_SWITCH_W("CD Capture Switch", 0, + MASK_ADDR_BURGUNDY_CAPTURESELECTS, 0, 16, 1), + BURGUNDY_SWITCH_W("CD Playback Switch", 0, + MASK_ADDR_BURGUNDY_OUTPUTSELECTS, 0, 16, 1), +/* BURGUNDY_SWITCH_W("Loop Capture Switch", 0, + * MASK_ADDR_BURGUNDY_CAPTURESELECTS, 8, 24, 1), + * BURGUNDY_SWITCH_B("Mixer out Capture Switch", 0, + * MASK_ADDR_BURGUNDY_HOSTIFAD, 0x02, 0, 0), + * BURGUNDY_SWITCH_B("Mixer Capture Switch", 0, + * MASK_ADDR_BURGUNDY_HOSTIFAD, 0x01, 0, 0), + * BURGUNDY_SWITCH_B("PCM out Capture Switch", 0, + * MASK_ADDR_BURGUNDY_HOSTIFEH, 0x02, 0, 0), + */ BURGUNDY_SWITCH_B("PCM Capture Switch", 0, + MASK_ADDR_BURGUNDY_HOSTIFEH, 0x01, 0, 0) +}; +static struct snd_kcontrol_new snd_pmac_burgundy_mixers_imac[] __initdata = { + BURGUNDY_VOLUME_W("Line in Capture Volume", 0, + MASK_ADDR_BURGUNDY_VOLLINE, 16), + BURGUNDY_VOLUME_W("Mic Capture Volume", 0, + MASK_ADDR_BURGUNDY_VOLMIC, 16), + BURGUNDY_VOLUME_B("Line in Gain Capture Volume", 0, + MASK_ADDR_BURGUNDY_GAINLINE, 1, 0), + BURGUNDY_VOLUME_B("Mic Gain Capture Volume", 0, + MASK_ADDR_BURGUNDY_GAINMIC, 1, 0), + BURGUNDY_VOLUME_B("PC Speaker Playback Volume", 0, + MASK_ADDR_BURGUNDY_ATTENSPEAKER, 1, 1), + BURGUNDY_VOLUME_B("Line out Playback Volume", 0, + MASK_ADDR_BURGUNDY_ATTENLINEOUT, 1, 1), + BURGUNDY_VOLUME_B("Headphone Playback Volume", 0, + MASK_ADDR_BURGUNDY_ATTENHP, 1, 1), + BURGUNDY_SWITCH_W("Line in Capture Switch", 0, + MASK_ADDR_BURGUNDY_CAPTURESELECTS, 1, 17, 1), + BURGUNDY_SWITCH_W("Mic Capture Switch", 0, + MASK_ADDR_BURGUNDY_CAPTURESELECTS, 2, 18, 1), + BURGUNDY_SWITCH_W("Line in Playback Switch", 0, + MASK_ADDR_BURGUNDY_OUTPUTSELECTS, 1, 17, 1), + BURGUNDY_SWITCH_W("Mic Playback Switch", 0, + MASK_ADDR_BURGUNDY_OUTPUTSELECTS, 2, 18, 1), + BURGUNDY_SWITCH_B("Mic Boost Capture Switch", 0, + MASK_ADDR_BURGUNDY_INPBOOST, 0x40, 0x80, 1) +}; +static struct snd_kcontrol_new snd_pmac_burgundy_mixers_pmac[] __initdata = { + BURGUNDY_VOLUME_W("Line in Capture Volume", 0, + MASK_ADDR_BURGUNDY_VOLMIC, 16), + BURGUNDY_VOLUME_B("Line in Gain Capture Volume", 0, + MASK_ADDR_BURGUNDY_GAINMIC, 1, 0), + BURGUNDY_VOLUME_B("PC Speaker Playback Volume", 0, + MASK_ADDR_BURGUNDY_ATTENMONO, 0, 1), + BURGUNDY_VOLUME_B("Line out Playback Volume", 0, + MASK_ADDR_BURGUNDY_ATTENSPEAKER, 1, 1), + BURGUNDY_SWITCH_W("Line in Capture Switch", 0, + MASK_ADDR_BURGUNDY_CAPTURESELECTS, 2, 18, 1), + BURGUNDY_SWITCH_W("Line in Playback Switch", 0, + MASK_ADDR_BURGUNDY_OUTPUTSELECTS, 2, 18, 1), +/* BURGUNDY_SWITCH_B("Line in Boost Capture Switch", 0, + * MASK_ADDR_BURGUNDY_INPBOOST, 0x40, 0x80, 1) */ +}; +static struct snd_kcontrol_new snd_pmac_burgundy_master_sw_imac __initdata = +BURGUNDY_SWITCH_B("Master Playback Switch", 0, + MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, + BURGUNDY_OUTPUT_LEFT | BURGUNDY_LINEOUT_LEFT | BURGUNDY_HP_LEFT, + BURGUNDY_OUTPUT_RIGHT | BURGUNDY_LINEOUT_RIGHT | BURGUNDY_HP_RIGHT, 1); +static struct snd_kcontrol_new snd_pmac_burgundy_master_sw_pmac __initdata = +BURGUNDY_SWITCH_B("Master Playback Switch", 0, + MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, + BURGUNDY_OUTPUT_INTERN + | BURGUNDY_OUTPUT_LEFT, BURGUNDY_OUTPUT_RIGHT, 1); +static struct snd_kcontrol_new snd_pmac_burgundy_speaker_sw_imac __initdata = +BURGUNDY_SWITCH_B("PC Speaker Playback Switch", 0, + MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, + BURGUNDY_OUTPUT_LEFT, BURGUNDY_OUTPUT_RIGHT, 1); +static struct snd_kcontrol_new snd_pmac_burgundy_speaker_sw_pmac __initdata = +BURGUNDY_SWITCH_B("PC Speaker Playback Switch", 0, + MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, + BURGUNDY_OUTPUT_INTERN, 0, 0); +static struct snd_kcontrol_new snd_pmac_burgundy_line_sw_imac __initdata = +BURGUNDY_SWITCH_B("Line out Playback Switch", 0, + MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, + BURGUNDY_LINEOUT_LEFT, BURGUNDY_LINEOUT_RIGHT, 1); +static struct snd_kcontrol_new snd_pmac_burgundy_line_sw_pmac __initdata = +BURGUNDY_SWITCH_B("Line out Playback Switch", 0, + MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, + BURGUNDY_OUTPUT_LEFT, BURGUNDY_OUTPUT_RIGHT, 1); +static struct snd_kcontrol_new snd_pmac_burgundy_hp_sw_imac __initdata = +BURGUNDY_SWITCH_B("Headphone Playback Switch", 0, + MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, + BURGUNDY_HP_LEFT, BURGUNDY_HP_RIGHT, 1); #ifdef PMAC_SUPPORT_AUTOMUTE @@ -350,16 +583,26 @@ static int snd_pmac_burgundy_detect_head static void snd_pmac_burgundy_update_automute(struct snd_pmac *chip, int do_notify) { if (chip->auto_mute) { + int imac = machine_is_compatible("iMac"); int reg, oreg; - reg = oreg = snd_pmac_burgundy_rcb(chip, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES); - reg &= ~(BURGUNDY_OUTPUT_LEFT | BURGUNDY_OUTPUT_RIGHT | BURGUNDY_OUTPUT_INTERN); + reg = oreg = snd_pmac_burgundy_rcb(chip, + MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES); + reg &= imac ? ~(BURGUNDY_OUTPUT_LEFT | BURGUNDY_OUTPUT_RIGHT + | BURGUNDY_HP_LEFT | BURGUNDY_HP_RIGHT) + : ~(BURGUNDY_OUTPUT_LEFT | BURGUNDY_OUTPUT_RIGHT + | BURGUNDY_OUTPUT_INTERN); if (snd_pmac_burgundy_detect_headphone(chip)) - reg |= BURGUNDY_OUTPUT_LEFT | BURGUNDY_OUTPUT_RIGHT; + reg |= imac ? (BURGUNDY_HP_LEFT | BURGUNDY_HP_RIGHT) + : (BURGUNDY_OUTPUT_LEFT + | BURGUNDY_OUTPUT_RIGHT); else - reg |= BURGUNDY_OUTPUT_INTERN; + reg |= imac ? (BURGUNDY_OUTPUT_LEFT + | BURGUNDY_OUTPUT_RIGHT) + : (BURGUNDY_OUTPUT_INTERN); if (do_notify && reg == oreg) return; - snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, reg); + snd_pmac_burgundy_wcb(chip, + MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, reg); if (do_notify) { snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->master_sw_ctl->id); @@ -378,6 +621,7 @@ static void snd_pmac_burgundy_update_aut */ int __init snd_pmac_burgundy_init(struct snd_pmac *chip) { + int imac = machine_is_compatible("iMac"); int i, err; /* Checks to see the chip is alive and kicking */ @@ -386,7 +630,7 @@ int __init snd_pmac_burgundy_init(struct return 1; } - snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_OUTPUTENABLES, + snd_pmac_burgundy_wcw(chip, MASK_ADDR_BURGUNDY_OUTPUTENABLES, DEF_BURGUNDY_OUTPUTENABLES); snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, DEF_BURGUNDY_MORE_OUTPUTENABLES); @@ -396,7 +640,8 @@ int __init snd_pmac_burgundy_init(struct snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_INPSEL21, DEF_BURGUNDY_INPSEL21); snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_INPSEL3, - DEF_BURGUNDY_INPSEL3); + imac ? DEF_BURGUNDY_INPSEL3_IMAC + : DEF_BURGUNDY_INPSEL3_PMAC); snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_GAINCD, DEF_BURGUNDY_GAINCD); snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_GAINLINE, @@ -422,23 +667,51 @@ int __init snd_pmac_burgundy_init(struct snd_pmac_burgundy_wcw(chip, MASK_ADDR_BURGUNDY_VOLMIC, DEF_BURGUNDY_VOLMIC); - if (chip->hp_stat_mask == 0) + if (chip->hp_stat_mask == 0) { /* set headphone-jack detection bit */ - chip->hp_stat_mask = 0x04; - + if (imac) + chip->hp_stat_mask = BURGUNDY_HPDETECT_IMAC_UPPER + | BURGUNDY_HPDETECT_IMAC_LOWER + | BURGUNDY_HPDETECT_IMAC_SIDE; + else + chip->hp_stat_mask = BURGUNDY_HPDETECT_PMAC_BACK; + } /* * build burgundy mixers */ strcpy(chip->card->mixername, "PowerMac Burgundy"); for (i = 0; i < ARRAY_SIZE(snd_pmac_burgundy_mixers); i++) { - if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_pmac_burgundy_mixers[i], chip))) < 0) + if ((err = snd_ctl_add(chip->card, + snd_ctl_new1(&snd_pmac_burgundy_mixers[i], chip))) < 0) + return err; + } + for (i = 0; i < (imac ? ARRAY_SIZE(snd_pmac_burgundy_mixers_imac) + : ARRAY_SIZE(snd_pmac_burgundy_mixers_pmac)); i++) { + if ((err = snd_ctl_add(chip->card, + snd_ctl_new1(imac ? &snd_pmac_burgundy_mixers_imac[i] + : &snd_pmac_burgundy_mixers_pmac[i], chip))) < 0) return err; } - chip->master_sw_ctl = snd_ctl_new1(&snd_pmac_burgundy_master_sw, chip); + chip->master_sw_ctl = snd_ctl_new1(imac + ? &snd_pmac_burgundy_master_sw_imac + : &snd_pmac_burgundy_master_sw_pmac, chip); if ((err = snd_ctl_add(chip->card, chip->master_sw_ctl)) < 0) return err; - chip->speaker_sw_ctl = snd_ctl_new1(&snd_pmac_burgundy_speaker_sw, chip); + chip->master_sw_ctl = snd_ctl_new1(imac + ? &snd_pmac_burgundy_line_sw_imac + : &snd_pmac_burgundy_line_sw_pmac, chip); + if ((err = snd_ctl_add(chip->card, chip->master_sw_ctl)) < 0) + return err; + if (imac) { + chip->master_sw_ctl = snd_ctl_new1( + &snd_pmac_burgundy_hp_sw_imac, chip); + if ((err = snd_ctl_add(chip->card, chip->master_sw_ctl)) < 0) + return err; + } + chip->speaker_sw_ctl = snd_ctl_new1(imac + ? &snd_pmac_burgundy_speaker_sw_imac + : &snd_pmac_burgundy_speaker_sw_pmac, chip); if ((err = snd_ctl_add(chip->card, chip->speaker_sw_ctl)) < 0) return err; #ifdef PMAC_SUPPORT_AUTOMUTE --- sound/ppc/burgundy.h.orig 2008-04-05 17:45:08.000000000 +0300 +++ sound/ppc/burgundy.h 2008-04-05 22:02:44.000000000 +0300 @@ -22,6 +22,7 @@ #ifndef __BURGUNDY_H #define __BURGUNDY_H +#define MASK_ADDR_BURGUNDY_INPBOOST (0x10 << 12) #define MASK_ADDR_BURGUNDY_INPSEL21 (0x11 << 12) #define MASK_ADDR_BURGUNDY_INPSEL3 (0x12 << 12) @@ -35,7 +36,10 @@ #define MASK_ADDR_BURGUNDY_VOLCH3 (0x22 << 12) #define MASK_ADDR_BURGUNDY_VOLCH4 (0x23 << 12) +#define MASK_ADDR_BURGUNDY_CAPTURESELECTS (0x2A << 12) #define MASK_ADDR_BURGUNDY_OUTPUTSELECTS (0x2B << 12) +#define MASK_ADDR_BURGUNDY_VOLMIX01 (0x2D << 12) +#define MASK_ADDR_BURGUNDY_VOLMIX23 (0x2E << 12) #define MASK_ADDR_BURGUNDY_OUTPUTENABLES (0x2F << 12) #define MASK_ADDR_BURGUNDY_MASTER_VOLUME (0x30 << 12) @@ -45,6 +49,10 @@ #define MASK_ADDR_BURGUNDY_ATTENSPEAKER (0x62 << 12) #define MASK_ADDR_BURGUNDY_ATTENLINEOUT (0x63 << 12) #define MASK_ADDR_BURGUNDY_ATTENHP (0x64 << 12) +#define MASK_ADDR_BURGUNDY_ATTENMONO (0x65 << 12) + +#define MASK_ADDR_BURGUNDY_HOSTIFAD (0x78 << 12) +#define MASK_ADDR_BURGUNDY_HOSTIFEH (0x79 << 12) #define MASK_ADDR_BURGUNDY_VOLCD (MASK_ADDR_BURGUNDY_VOLCH1) #define MASK_ADDR_BURGUNDY_VOLLINE (MASK_ADDR_BURGUNDY_VOLCH2) @@ -59,21 +67,22 @@ /* These are all default values for the burgundy */ #define DEF_BURGUNDY_INPSEL21 (0xAA) -#define DEF_BURGUNDY_INPSEL3 (0x0A) +#define DEF_BURGUNDY_INPSEL3_IMAC (0x0A) +#define DEF_BURGUNDY_INPSEL3_PMAC (0x05) #define DEF_BURGUNDY_GAINCD (0x33) #define DEF_BURGUNDY_GAINLINE (0x44) #define DEF_BURGUNDY_GAINMIC (0x44) #define DEF_BURGUNDY_GAINMODEM (0x06) -/* Remember: lowest volume here is 0x9b */ +/* Remember: lowest volume here is 0x9B (155) */ #define DEF_BURGUNDY_VOLCD (0xCCCCCCCC) #define DEF_BURGUNDY_VOLLINE (0x00000000) #define DEF_BURGUNDY_VOLMIC (0x00000000) #define DEF_BURGUNDY_VOLMODEM (0xCCCCCCCC) -#define DEF_BURGUNDY_OUTPUTSELECTS (0x010f010f) -#define DEF_BURGUNDY_OUTPUTENABLES (0x0A) +#define DEF_BURGUNDY_OUTPUTSELECTS (0x010F010F) +#define DEF_BURGUNDY_OUTPUTENABLES (0x0100000A) /* #define DEF_BURGUNDY_MASTER_VOLUME (0xFFFFFFFF) */ /* too loud */ #define DEF_BURGUNDY_MASTER_VOLUME (0xDDDDDDDD) @@ -84,12 +93,22 @@ #define DEF_BURGUNDY_ATTENLINEOUT (0xCC) #define DEF_BURGUNDY_ATTENHP (0xCC) -/* OUTPUTENABLES bits */ +/* MORE_OUTPUTENABLES bits */ #define BURGUNDY_OUTPUT_LEFT 0x02 #define BURGUNDY_OUTPUT_RIGHT 0x04 +#define BURGUNDY_LINEOUT_LEFT 0x08 +#define BURGUNDY_LINEOUT_RIGHT 0x10 +#define BURGUNDY_HP_LEFT 0x20 +#define BURGUNDY_HP_RIGHT 0x40 #define BURGUNDY_OUTPUT_INTERN 0x80 -/* volume offset */ +/* Headphone detection bits */ +#define BURGUNDY_HPDETECT_PMAC_BACK 0x04 +#define BURGUNDY_HPDETECT_IMAC_SIDE 0x04 +#define BURGUNDY_HPDETECT_IMAC_UPPER 0x08 +#define BURGUNDY_HPDETECT_IMAC_LOWER 0x01 + +/* Volume offset */ #define BURGUNDY_VOLUME_OFFSET 155 #endif /* __BURGUNDY_H */

1 2

[alsa-devel] [PATCH 002b/003] snd-powermac: style awacs.s and awacs.h
by Risto Suominen 16 Apr '08

16 Apr '08

From: Risto Suominen <Risto.Suominen(a)gmail.com> Coding style corrections for awacs.c and awacs.h. Signed-off-by: Risto Suominen <Risto.Suominen(a)gmail.com> --- Kernel version 2.6.25-rc8 (probably any 2.6) --- sound/ppc/awacs.c.orig 2008-04-05 17:45:08.000000000 +0300 +++ sound/ppc/awacs.c 2008-04-07 11:24:08.108621048 +0300 @@ -141,7 +141,7 @@ static int snd_pmac_awacs_info_volume(st uinfo->value.integer.max = 15; return 0; } - + static int snd_pmac_awacs_get_volume(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -267,7 +267,8 @@ static int snd_pmac_awacs_put_switch(str static void awacs_set_cuda(int reg, int val) { struct adb_request req; - cuda_request(&req, NULL, 5, CUDA_PACKET, CUDA_GET_SET_IIC, 0x8a, reg, val); + cuda_request(&req, NULL, 5, CUDA_PACKET, CUDA_GET_SET_IIC, 0x8a, + reg, val); while (! req.complete) cuda_poll(); } @@ -289,11 +290,11 @@ static void awacs_amp_set_tone(struct aw /* * vol = 0 - 31 (attenuation), 32 = mute bit, stereo */ -static int awacs_amp_set_vol(struct awacs_amp *amp, int index, int lvol, int rvol, - int do_check) +static int awacs_amp_set_vol(struct awacs_amp *amp, int index, + int lvol, int rvol, int do_check) { if (do_check && amp->amp_vol[index][0] == lvol && - amp->amp_vol[index][1] == rvol) + amp->amp_vol[index][1] == rvol) return 0; awacs_set_cuda(3 + index, lvol); awacs_set_cuda(5 + index, rvol); @@ -337,7 +338,7 @@ static int snd_pmac_awacs_info_volume_am uinfo->value.integer.max = 31; return 0; } - + static int snd_pmac_awacs_get_volume_amp(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -361,8 +362,10 @@ static int snd_pmac_awacs_put_volume_amp snd_assert(amp, return -EINVAL); snd_assert(index >= 0 && index <= 1, return -EINVAL); - vol[0] = (31 - (ucontrol->value.integer.value[0] & 31)) | (amp->amp_vol[index][0] & 32); - vol[1] = (31 - (ucontrol->value.integer.value[1] & 31)) | (amp->amp_vol[index][1] & 32); + vol[0] = (31 - (ucontrol->value.integer.value[0] & 31)) + | (amp->amp_vol[index][0] & 32); + vol[1] = (31 - (ucontrol->value.integer.value[1] & 31)) + | (amp->amp_vol[index][1] & 32); return awacs_amp_set_vol(amp, index, vol[0], vol[1], 1); } @@ -374,8 +377,10 @@ static int snd_pmac_awacs_get_switch_amp struct awacs_amp *amp = chip->mixer_data; snd_assert(amp, return -EINVAL); snd_assert(index >= 0 && index <= 1, return -EINVAL); - ucontrol->value.integer.value[0] = (amp->amp_vol[index][0] & 32) ? 0 : 1; - ucontrol->value.integer.value[1] = (amp->amp_vol[index][1] & 32) ? 0 : 1; + ucontrol->value.integer.value[0] = (amp->amp_vol[index][0] & 32) + ? 0 : 1; + ucontrol->value.integer.value[1] = (amp->amp_vol[index][1] & 32) + ? 0 : 1; return 0; } @@ -389,8 +394,10 @@ static int snd_pmac_awacs_put_switch_amp snd_assert(amp, return -EINVAL); snd_assert(index >= 0 && index <= 1, return -EINVAL); - vol[0] = (ucontrol->value.integer.value[0] ? 0 : 32) | (amp->amp_vol[index][0] & 31); - vol[1] = (ucontrol->value.integer.value[1] ? 0 : 32) | (amp->amp_vol[index][1] & 31); + vol[0] = (ucontrol->value.integer.value[0] ? 0 : 32) + | (amp->amp_vol[index][0] & 31); + vol[1] = (ucontrol->value.integer.value[1] ? 0 : 32) + | (amp->amp_vol[index][1] & 31); return awacs_amp_set_vol(amp, index, vol[0], vol[1], 1); } @@ -403,7 +410,7 @@ static int snd_pmac_awacs_info_tone_amp( uinfo->value.integer.max = 14; return 0; } - + static int snd_pmac_awacs_get_tone_amp(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -445,7 +452,7 @@ static int snd_pmac_awacs_info_master_am uinfo->value.integer.max = 99; return 0; } - + static int snd_pmac_awacs_get_master_amp(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -688,12 +694,14 @@ static struct snd_kcontrol_new snd_pmac_ /* * add new mixer elements to the card */ -static int build_mixers(struct snd_pmac *chip, int nums, struct snd_kcontrol_new *mixers) +static int build_mixers(struct snd_pmac *chip, int nums, + struct snd_kcontrol_new *mixers) { int i, err; for (i = 0; i < nums; i++) { - if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&mixers[i], chip))) < 0) + if ((err = snd_ctl_add(chip->card, + snd_ctl_new1(&mixers[i], chip))) < 0) return err; } return 0; @@ -743,8 +752,10 @@ static void snd_pmac_awacs_resume(struct #ifdef PMAC_AMP_AVAIL if (chip->mixer_data) { struct awacs_amp *amp = chip->mixer_data; - awacs_amp_set_vol(amp, 0, amp->amp_vol[0][0], amp->amp_vol[0][1], 0); - awacs_amp_set_vol(amp, 1, amp->amp_vol[1][0], amp->amp_vol[1][1], 0); + awacs_amp_set_vol(amp, 0, + amp->amp_vol[0][0], amp->amp_vol[0][1], 0); + awacs_amp_set_vol(amp, 1, + amp->amp_vol[1][0], amp->amp_vol[1][1], 0); awacs_amp_set_tone(amp, amp->amp_tone[0], amp->amp_tone[1]); awacs_amp_set_master(amp, amp->amp_master); } @@ -849,7 +860,7 @@ snd_pmac_awacs_init(struct snd_pmac *chi chip->awacs_reg[1] = MASK_CMUTE | MASK_AMUTE; /* FIXME: Only machines with external SRS module need MASK_PAROUT */ if (chip->has_iic || chip->device_id == 0x5 || - /*chip->_device_id == 0x8 || */ + /* chip->_device_id == 0x8 || */ chip->device_id == 0xb) chip->awacs_reg[1] |= MASK_PAROUT; /* get default volume from nvram */ @@ -860,8 +871,10 @@ snd_pmac_awacs_init(struct snd_pmac *chi chip->awacs_reg[2] = vol; chip->awacs_reg[4] = vol; if (chip->model == PMAC_SCREAMER) { - chip->awacs_reg[5] = vol; /* FIXME: screamer has loopthru vol control */ - chip->awacs_reg[6] = MASK_MIC_BOOST; /* FIXME: maybe should be vol << 3 for PCMCIA speaker */ + /* FIXME: screamer has loopthru vol control */ + chip->awacs_reg[5] = vol; + /* FIXME: maybe should be vol << 3 for PCMCIA speaker */ + chip->awacs_reg[6] = MASK_MIC_BOOST; chip->awacs_reg[7] = 0; } @@ -877,7 +890,8 @@ snd_pmac_awacs_init(struct snd_pmac *chi return -ENOMEM; chip->mixer_data = amp; chip->mixer_free = awacs_amp_free; - awacs_amp_set_vol(amp, 0, 63, 63, 0); /* mute and zero vol */ + /* mute and zero vol */ + awacs_amp_set_vol(amp, 0, 63, 63, 0); awacs_amp_set_vol(amp, 1, 63, 63, 0); awacs_amp_set_tone(amp, 7, 7); /* 0 dB */ awacs_amp_set_master(amp, 79); /* 0 dB */ @@ -965,21 +979,25 @@ snd_pmac_awacs_init(struct snd_pmac *chi * screamer registers. * in this case, it seems the route C is not used. */ - if ((err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_amp_vol), + if ((err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_awacs_amp_vol), snd_pmac_awacs_amp_vol)) < 0) return err; /* overwrite */ - chip->master_sw_ctl = snd_ctl_new1(&snd_pmac_awacs_amp_hp_sw, chip); + chip->master_sw_ctl = snd_ctl_new1(&snd_pmac_awacs_amp_hp_sw, + chip); if ((err = snd_ctl_add(chip->card, chip->master_sw_ctl)) < 0) return err; - chip->speaker_sw_ctl = snd_ctl_new1(&snd_pmac_awacs_amp_spk_sw, chip); + chip->speaker_sw_ctl = snd_ctl_new1(&snd_pmac_awacs_amp_spk_sw, + chip); if ((err = snd_ctl_add(chip->card, chip->speaker_sw_ctl)) < 0) return err; } else #endif /* PMAC_AMP_AVAIL */ { /* route A = headphone, route C = speaker */ - if ((err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_speaker_vol), + if ((err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_awacs_speaker_vol), snd_pmac_awacs_speaker_vol)) < 0) return err; chip->speaker_sw_ctl = snd_ctl_new1(imac @@ -1010,8 +1028,9 @@ snd_pmac_awacs_init(struct snd_pmac *chi snd_pmac_awacs_mic_boost_pmac7500)) < 0) return err; } else { - if ((err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_mic_boost), - snd_pmac_awacs_mic_boost)) < 0) + if ((err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_awacs_mic_boost), + snd_pmac_awacs_mic_boost)) < 0) return err; } --- sound/ppc/awacs.h.orig 2008-04-05 17:45:08.000000000 +0300 +++ sound/ppc/awacs.h 2008-04-07 11:26:58.198763416 +0300 @@ -175,7 +175,8 @@ struct awacs_regs { /* DBDMA ChannelStatus Bit Masks */ /* ----- ------------- --- ----- */ #define MASK_CSERR (0x1 << 7) /* Error */ -#define MASK_EOI (0x1 << 6) /* End of Input -- only for Input Channel */ +#define MASK_EOI (0x1 << 6) /* End of Input -- + only for Input Channel */ #define MASK_CSUNUSED (0x1f << 1) /* bits 1-5 not used */ #define MASK_WAIT (0x1) /* Wait */

2 6

[alsa-devel] [PATCH 6/6] sound/drivers/dummy.c: fix negative snd_pcm_format_width() check
by Roel Kluin 16 Apr '08

16 Apr '08

bps is unsigned, a negative snd_pcm_format_width() return value is not noticed Signed-off-by: Roel Kluin <12o3l(a)tiscali.nl> --- diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index a240eae..1c88977 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -230,13 +230,14 @@ static int snd_card_dummy_pcm_prepare(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_dummy_pcm *dpcm = runtime->private_data; - unsigned int bps; + int bps; + + bps = snd_pcm_format_width(runtime->format) * runtime->rate * + runtime->channels / 8; - bps = runtime->rate * runtime->channels; - bps *= snd_pcm_format_width(runtime->format); - bps /= 8; if (bps <= 0) return -EINVAL; + dpcm->pcm_bps = bps; dpcm->pcm_jiffie = bps / HZ; dpcm->pcm_size = snd_pcm_lib_buffer_bytes(substream);

2 1

[alsa-devel] How to kill thread server_job() in snd_pcm_open()
by Rong-Jhe 16 Apr '08

16 Apr '08

I found that the thread server_job() can not be killed by using snd_pcm_close (). The code in snd_pcm_direct_server_discard() is disabled, and then it can not send signal to kill the thread. Why the code is disabled? Is there any bugs if I enable the code?

3 4

[alsa-devel] no sound with snd-hda-intel - error docu
by Ulrich Breymann 16 Apr '08

16 Apr '08

Dear Alsa-Developers, unfortunately I hear no sound at all with SuSE Linux 10.3. Volume is switched on (kmix). The hardware is ok, it functions with Windows. For your information I append - the content of /proc/asound/card0/codec#2 (file intelcodec.txt) - the result of command 'amixer -i' (file amixer_i.txt) - the result of command 'lspci -nv' (file lspci_nv.txt) and here the short results of some other commands: lspci | grep -i audio: 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 02) lsmod | grep hda_intel: snd_hda_intel 285212 1 snd_pcm 94852 2 snd_pcm_oss,snd_hda_intel snd 70452 9 snd_pcm_oss,snd_mixer_oss,snd_seq,snd_seq_device,snd_hda_intel,snd_pcm,snd_timer snd_page_alloc 26248 2 snd_hda_intel,snd_pcm rpm -qa | grep kernel: kernel-bigsmp-2.6.22.5-31 kernel-source-2.6.22.5-31 linux-kernel-headers-2.6.22-19 cat /etc/modprobe.d/sound: options snd-hda-intel enable=1 index=0 # u1Nb.Eq+LVzceV9F:82801I (ICH9 Family) HD Audio Controller alias snd-card-0 snd-hda-intel I hope this can help you (and later me & others). Best regards Uli content of /proc/asound/card0/codec#2 : Codec: SigmaTel STAC9274D Address: 2 Vendor Id: 0x83847621 Subsystem Id: 0x80860519 Revision Id: 0x100201 No Modem Function Group found Default PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: ofs=0x00, nsteps=0x0e, stepsize=0x05, mute=0 Default Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x02, mute=1 Node 0x02 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x57 0x57] Power: 0x0 Node 0x03 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x2b 0x2b] Power: 0x0 Node 0x04 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x2b 0x49] Power: 0x0 Node 0x05 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x1f 0x1f] Power: 0x0 Node 0x06 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Power: 0x0 Node 0x07 [Audio Input] wcaps 0x1d0541: Stereo Power: 0x0 Connection: 1 0x1b Node 0x08 [Audio Input] wcaps 0x1d0541: Stereo Power: 0x0 Connection: 1 0x1c Node 0x09 [Audio Input] wcaps 0x1d0541: Stereo Power: 0x0 Connection: 1 0x1d Node 0x0a [Pin Complex] wcaps 0x400181: Stereo Pincap 0x08173f: IN OUT HP Detect Pin Default 0x0221401f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green Pin-ctls: 0xc0: OUT HP Connection: 3 0x02* 0x03 0x06 Node 0x0b [Pin Complex] wcaps 0x400181: Stereo Pincap 0x08173f: IN OUT HP Detect Pin Default 0x02a1902e: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink Pin-ctls: 0x24: IN Connection: 3 0x02* 0x03 0x06 Node 0x0c [Pin Complex] wcaps 0x400181: Stereo Pincap 0x081737: IN OUT Detect Pin Default 0x01813021: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue Pin-ctls: 0x20: IN Connection: 1 0x03 Node 0x0d [Pin Complex] wcaps 0x400181: Stereo Pincap 0x08173f: IN OUT HP Detect Pin Default 0x01114010: [Jack] Speaker at Ext Rear Conn = 1/8, Color = Green Pin-ctls: 0x40: OUT Connection: 1 0x02 Node 0x0e [Pin Complex] wcaps 0x400181: Stereo Pincap 0x081737: IN OUT Detect Pin Default 0x01a19020: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink Pin-ctls: 0x24: IN Connection: 1 0x04 Node 0x0f [Pin Complex] wcaps 0x400181: Stereo Pincap 0x081737: IN OUT Detect Pin Default 0x01111012: [Jack] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x40: OUT Connection: 1 0x05 Node 0x10 [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0837: IN OUT Detect Pin Default 0x01117011: [Jack] Speaker at Ext Rear Conn = 1/8, Color = Yellow Pin-ctls: 0x40: OUT Connection: 1 0x04 Node 0x11 [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0837: IN OUT Detect Pin Default 0x400000fb: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Connection: 1 0x03 Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x500000fe: [N/A] Line Out at Int N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Node 0x13 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x500001ff: [N/A] Line Out at Int N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Node 0x14 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x500001fd: [N/A] Line Out at Int N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Node 0x15 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 9 0x0e* 0x12 0x0f 0x0b 0x0c 0x0d 0x0a 0x10 0x11 Node 0x16 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 9 0x0e* 0x12 0x0f 0x0b 0x0c 0x0d 0x0a 0x10 0x11 Node 0x17 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 9 0x0e* 0x12 0x0f 0x0b 0x0c 0x0d 0x0a 0x10 0x11 Node 0x18 [Audio Selector] wcaps 0x300103: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Connection: 1 0x15 Node 0x19 [Audio Selector] wcaps 0x300103: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Connection: 1 0x16 Node 0x1a [Audio Selector] wcaps 0x300103: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Connection: 1 0x17 Node 0x1b [Audio Selector] wcaps 0x30090d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Connection: 3 0x18* 0x13 0x14 Node 0x1c [Audio Selector] wcaps 0x30090d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Connection: 3 0x19* 0x13 0x14 Node 0x1d [Audio Selector] wcaps 0x30090d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Connection: 3 0x1a* 0x13 0x14 Node 0x1e [Audio Output] wcaps 0x40211: Stereo Digital PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Node 0x1f [Vendor Defined Widget] wcaps 0xf30201: Stereo Digital Node 0x20 [Audio Input] wcaps 0x140311: Stereo Digital PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Connection: 1 0x22 Node 0x21 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x01452130: [Jack] SPDIF Out at Ext Rear Conn = Optical, Color = Grey Pin-ctls: 0x40: OUT Connection: 5 0x1e* 0x1f 0x1b 0x1c 0x1d Node 0x22 [Pin Complex] wcaps 0x430681: Stereo Digital Pincap 0x0810024: IN EAPD Detect Pin Default 0x400000fc: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Power: 0x0 Node 0x23 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=0 Amp-Out vals: [0x00] Node 0x24 [Volume Knob Widget] wcaps 0x600000: Mono Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 122 [48%] [-26.60dB] Front Right: Playback 122 [48%] [-26.60dB] Simple mixer control 'Front',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 87 [69%] [-30.00dB] [on] Front Right: Playback 87 [69%] [-30.00dB] [on] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 31 [24%] [-72.00dB] [on] Front Right: Playback 31 [24%] [-72.00dB] [on] Simple mixer control 'Center',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 127 Mono: Playback 43 [34%] [-63.00dB] [on] Simple mixer control 'LFE',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 127 Mono: Playback 73 [57%] [-40.50dB] [on] Simple mixer control 'Side',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 43 [34%] [-63.00dB] [on] Front Right: Playback 43 [34%] [-63.00dB] [on] Simple mixer control 'Line In as Output',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'ADCMux',0 Capabilities: cswitch Capture channels: Front Left - Front Right Front Left: Capture [on] Front Right: Capture [on] Simple mixer control 'InMux',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 4 Front Left: Capture 0 [0%] [0.00dB] Front Right: Capture 0 [0%] [0.00dB] Simple mixer control 'InVol',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 14 Front Left: Capture 0 [0%] [0.00dB] Front Right: Capture 0 [0%] [0.00dB] Simple mixer control 'Input Source',0 Capabilities: cenum Items: 'Mic' 'Front Mic' 'Line' Item0: 'Mic' 00:00.0 Class 0600: 8086:29e0 (rev 01) Subsystem: 8086:5442 Flags: bus master, fast devsel, latency 0 Capabilities: [e0] Vendor Specific Information 00:01.0 Class 0604: 8086:29e1 (rev 01) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=01, subordinate=01, sec-latency=0 I/O behind bridge: 00002000-00002fff Memory behind bridge: e0000000-e2ffffff Prefetchable memory behind bridge: 00000000d0000000-00000000dfffffff Capabilities: [88] Subsystem: 8086:5442 Capabilities: [80] Power Management version 3 Capabilities: [90] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable+ Capabilities: [a0] Express Root Port (Slot+) IRQ 0 Capabilities: [100] Virtual Channel Capabilities: [140] Unknown (5) 00:19.0 Class 0200: 8086:294c (rev 02) Subsystem: 8086:0001 Flags: bus master, fast devsel, latency 0, IRQ 220 Memory at e3200000 (32-bit, non-prefetchable) [size=128K] Memory at e3224000 (32-bit, non-prefetchable) [size=4K] I/O ports at 30e0 [size=32] Capabilities: [c8] Power Management version 2 Capabilities: [d0] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable+ Capabilities: [e0] Vendor Specific Information 00:1a.0 Class 0c03: 8086:2937 (rev 02) (prog-if 00 [UHCI]) Subsystem: 8086:5442 Flags: bus master, medium devsel, latency 0, IRQ 19 I/O ports at 30c0 [size=32] Capabilities: [50] Vendor Specific Information 00:1a.1 Class 0c03: 8086:2938 (rev 02) (prog-if 00 [UHCI]) Subsystem: 8086:5442 Flags: bus master, medium devsel, latency 0, IRQ 18 I/O ports at 30a0 [size=32] Capabilities: [50] Vendor Specific Information 00:1a.2 Class 0c03: 8086:2939 (rev 02) (prog-if 00 [UHCI]) Subsystem: 8086:5442 Flags: bus master, medium devsel, latency 0, IRQ 17 I/O ports at 3080 [size=32] Capabilities: [50] Vendor Specific Information 00:1a.7 Class 0c03: 8086:293c (rev 02) (prog-if 20 [EHCI]) Subsystem: 8086:5442 Flags: bus master, medium devsel, latency 0, IRQ 17 Memory at e3225400 (32-bit, non-prefetchable) [size=1K] Capabilities: [50] Power Management version 2 Capabilities: [58] Debug port Capabilities: [98] Vendor Specific Information 00:1b.0 Class 0403: 8086:293e (rev 02) Subsystem: 8086:0519 Flags: bus master, fast devsel, latency 0, IRQ 23 Memory at e3220000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) 00:1c.0 Class 0604: 8086:2940 (rev 02) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=02, subordinate=02, sec-latency=0 Memory behind bridge: e3300000-e33fffff Capabilities: [40] Express Root Port (Slot+) IRQ 0 Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable+ Capabilities: [90] Subsystem: 8086:2940 Capabilities: [a0] Power Management version 2 Capabilities: [100] Virtual Channel Capabilities: [180] Unknown (5) 00:1c.4 Class 0604: 8086:2948 (rev 02) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=03, subordinate=03, sec-latency=0 I/O behind bridge: 00001000-00001fff Memory behind bridge: e3100000-e31fffff Capabilities: [40] Express Root Port (Slot+) IRQ 0 Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable+ Capabilities: [90] Subsystem: 8086:2948 Capabilities: [a0] Power Management version 2 Capabilities: [100] Virtual Channel Capabilities: [180] Unknown (5) 00:1d.0 Class 0c03: 8086:2934 (rev 02) (prog-if 00 [UHCI]) Subsystem: 8086:5442 Flags: bus master, medium devsel, latency 0, IRQ 20 I/O ports at 3060 [size=32] Capabilities: [50] Vendor Specific Information 00:1d.1 Class 0c03: 8086:2935 (rev 02) (prog-if 00 [UHCI]) Subsystem: 8086:5442 Flags: bus master, medium devsel, latency 0, IRQ 21 I/O ports at 3040 [size=32] Capabilities: [50] Vendor Specific Information 00:1d.2 Class 0c03: 8086:2936 (rev 02) (prog-if 00 [UHCI]) Subsystem: 8086:5442 Flags: bus master, medium devsel, latency 0, IRQ 19 I/O ports at 3020 [size=32] Capabilities: [50] Vendor Specific Information 00:1d.7 Class 0c03: 8086:293a (rev 02) (prog-if 20 [EHCI]) Subsystem: 8086:5442 Flags: bus master, medium devsel, latency 0, IRQ 20 Memory at e3225000 (32-bit, non-prefetchable) [size=1K] Capabilities: [50] Power Management version 2 Capabilities: [58] Debug port Capabilities: [98] Vendor Specific Information 00:1e.0 Class 0604: 8086:244e (rev 92) (prog-if 01 [Subtractive decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=04, subordinate=04, sec-latency=32 Memory behind bridge: e3000000-e30fffff Capabilities: [50] Subsystem: 8086:5442 00:1f.0 Class 0601: 8086:2916 (rev 02) Subsystem: 8086:5442 Flags: bus master, medium devsel, latency 0 Capabilities: [e0] Vendor Specific Information 00:1f.2 Class 0101: 8086:2920 (rev 02) (prog-if 8f [Master SecP SecO PriP PriO]) Subsystem: 8086:5442 Flags: bus master, 66MHz, medium devsel, latency 0, IRQ 18 I/O ports at 3458 [size=8] I/O ports at 346c [size=4] I/O ports at 3450 [size=8] I/O ports at 3468 [size=4] I/O ports at 3430 [size=16] I/O ports at 3420 [size=16] Capabilities: [70] Power Management version 3 Capabilities: [b0] Vendor Specific Information 00:1f.3 Class 0c05: 8086:2930 (rev 02) Subsystem: 8086:5442 Flags: medium devsel, IRQ 19 Memory at e3225800 (64-bit, non-prefetchable) [size=256] I/O ports at 3000 [size=32] 00:1f.5 Class 0101: 8086:2926 (rev 02) (prog-if 85 [Master SecO PriO]) Subsystem: 8086:5442 Flags: bus master, 66MHz, medium devsel, latency 0, IRQ 18 I/O ports at 3448 [size=8] I/O ports at 3464 [size=4] I/O ports at 3440 [size=8] I/O ports at 3460 [size=4] I/O ports at 3410 [size=16] I/O ports at 3400 [size=16] Capabilities: [70] Power Management version 3 Capabilities: [b0] Vendor Specific Information 01:00.0 Class 0300: 10de:0402 (rev a1) (prog-if 00 [VGA]) Subsystem: 1458:3447 Flags: bus master, fast devsel, latency 0, IRQ 16 Memory at e2000000 (32-bit, non-prefetchable) [size=16M] Memory at d0000000 (64-bit, prefetchable) [size=256M] Memory at e0000000 (64-bit, non-prefetchable) [size=32M] I/O ports at 2000 [size=128] Capabilities: [60] Power Management version 2 Capabilities: [68] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [78] Express Endpoint IRQ 0 Capabilities: [100] Virtual Channel Capabilities: [128] Power Budgeting Capabilities: [600] Unknown (11) 03:00.0 Class 0101: 11ab:6121 (rev b2) (prog-if 8f [Master SecP SecO PriP PriO]) Subsystem: 8086:5442 Flags: bus master, fast devsel, latency 0, IRQ 16 I/O ports at 1018 [size=8] I/O ports at 1024 [size=4] I/O ports at 1010 [size=8] I/O ports at 1020 [size=4] I/O ports at 1000 [size=16] Memory at e3100000 (32-bit, non-prefetchable) [size=1K] Capabilities: [48] Power Management version 2 Capabilities: [50] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable- Capabilities: [e0] Express Legacy Endpoint IRQ 0 Capabilities: [100] Advanced Error Reporting 04:03.0 Class 0c00: 104c:8023 (prog-if 10 [OHCI]) Subsystem: 8086:5442 Flags: bus master, medium devsel, latency 32, IRQ 21 Memory at e3004000 (32-bit, non-prefetchable) [size=2K] Memory at e3000000 (32-bit, non-prefetchable) [size=16K] Capabilities: [44] Power Management version 2

2 1

[alsa-devel] [PATCH] soc - wm9712: Remove unneeded AC97_EXTENDED_MID updates
by Mark Brown 16 Apr '08

16 Apr '08

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9712.c | 8 -------- 1 files changed, 0 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 524f745..d2d79e1 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -581,22 +581,14 @@ static int wm9712_dapm_event(struct snd_soc_codec *codec, int event) switch (event) { case SNDRV_CTL_POWER_D0: /* full On */ - /* liam - maybe enable thermal shutdown */ - reg = ac97_read(codec, AC97_EXTENDED_MID) & 0xdfff; - ac97_write(codec, AC97_EXTENDED_MID, reg); - break; case SNDRV_CTL_POWER_D1: /* partial On */ case SNDRV_CTL_POWER_D2: /* partial On */ break; case SNDRV_CTL_POWER_D3hot: /* Off, with power */ - /* enable master bias and vmid */ - reg = ac97_read(codec, AC97_EXTENDED_MID) & 0xbbff; - ac97_write(codec, AC97_EXTENDED_MID, reg); ac97_write(codec, AC97_POWERDOWN, 0x0000); break; case SNDRV_CTL_POWER_D3cold: /* Off, without power */ /* disable everything including AC link */ - ac97_write(codec, AC97_EXTENDED_MID, 0xffff); ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); ac97_write(codec, AC97_POWERDOWN, 0xffff); break; -- 1.5.5

2 1

[alsa-devel] Releasing IRQs in older kernels
by Risto Suominen 16 Apr '08

16 Apr '08

I'd like to bring up a problem with using alsa-driver on older kernels. I already made a bug report regarding snd-powermac: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3883. It's possible that other drivers suffer from this, too. If I've understood it right, 1.0.16 should be usable on 2.2 kernels and newer. Well, it woudn't compile on 2.6.15-26-powerpc (Ubuntu 6.06.1 Dapper) without some patching. However, the real problem began, when I tried to unload the module. It seems that something has changed (when?) in the kernel's IRQ interface, and ALSA tries to make old code compatible with it by introducing snd_request/free_irq, to be used with older kernels. The idea is to distuingish between IRQs by the last parameter, usually chip's address. This leads to trouble when several IRQs come from the same chip, as is the case with snd-powermac. My conclusion: the drivers suffering from this should be identified, and some alternative way of handling IRQs correctly on different kernel versions should be developed. Risto

2 3

[alsa-devel] [PATCH 002/003] snd-powermac: AWACS and Screamer mixers for PM7500, Beige, and iMac SL
by Risto Suominen 16 Apr '08

16 Apr '08

From: Risto Suominen <Risto.Suominen at gmail.com> Add mixer controls and correct headphone detection bits for PowerMacs 7300/7500 (AWACS) and G3 Beige (Screamer), and iMac G3 Slot-loading (Screamer). Signed-off-by: Risto Suominen <Risto.Suominen at gmail.com> --- Kernel version 2.6.25-rc8 (probably any 2.6) --- sound/ppc/awacs.c.orig 2008-04-05 17:45:08.000000000 +0300 +++ sound/ppc/awacs.c 2008-04-07 11:24:08.108621048 +0300 @@ -141,7 +141,7 @@ static int snd_pmac_awacs_info_volume(st uinfo->value.integer.max = 15; return 0; } - + static int snd_pmac_awacs_get_volume(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -267,7 +267,8 @@ static int snd_pmac_awacs_put_switch(str static void awacs_set_cuda(int reg, int val) { struct adb_request req; - cuda_request(&req, NULL, 5, CUDA_PACKET, CUDA_GET_SET_IIC, 0x8a, reg, val); + cuda_request(&req, NULL, 5, CUDA_PACKET, CUDA_GET_SET_IIC, 0x8a, + reg, val); while (! req.complete) cuda_poll(); } @@ -289,11 +290,11 @@ static void awacs_amp_set_tone(struct aw /* * vol = 0 - 31 (attenuation), 32 = mute bit, stereo */ -static int awacs_amp_set_vol(struct awacs_amp *amp, int index, int lvol, int rvol, - int do_check) +static int awacs_amp_set_vol(struct awacs_amp *amp, int index, + int lvol, int rvol, int do_check) { if (do_check && amp->amp_vol[index][0] == lvol && - amp->amp_vol[index][1] == rvol) + amp->amp_vol[index][1] == rvol) return 0; awacs_set_cuda(3 + index, lvol); awacs_set_cuda(5 + index, rvol); @@ -337,7 +338,7 @@ static int snd_pmac_awacs_info_volume_am uinfo->value.integer.max = 31; return 0; } - + static int snd_pmac_awacs_get_volume_amp(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -361,8 +362,10 @@ static int snd_pmac_awacs_put_volume_amp snd_assert(amp, return -EINVAL); snd_assert(index >= 0 && index <= 1, return -EINVAL); - vol[0] = (31 - (ucontrol->value.integer.value[0] & 31)) | (amp->amp_vol[index][0] & 32); - vol[1] = (31 - (ucontrol->value.integer.value[1] & 31)) | (amp->amp_vol[index][1] & 32); + vol[0] = (31 - (ucontrol->value.integer.value[0] & 31)) + | (amp->amp_vol[index][0] & 32); + vol[1] = (31 - (ucontrol->value.integer.value[1] & 31)) + | (amp->amp_vol[index][1] & 32); return awacs_amp_set_vol(amp, index, vol[0], vol[1], 1); } @@ -374,8 +377,10 @@ static int snd_pmac_awacs_get_switch_amp struct awacs_amp *amp = chip->mixer_data; snd_assert(amp, return -EINVAL); snd_assert(index >= 0 && index <= 1, return -EINVAL); - ucontrol->value.integer.value[0] = (amp->amp_vol[index][0] & 32) ? 0 : 1; - ucontrol->value.integer.value[1] = (amp->amp_vol[index][1] & 32) ? 0 : 1; + ucontrol->value.integer.value[0] = (amp->amp_vol[index][0] & 32) + ? 0 : 1; + ucontrol->value.integer.value[1] = (amp->amp_vol[index][1] & 32) + ? 0 : 1; return 0; } @@ -389,8 +394,10 @@ static int snd_pmac_awacs_put_switch_amp snd_assert(amp, return -EINVAL); snd_assert(index >= 0 && index <= 1, return -EINVAL); - vol[0] = (ucontrol->value.integer.value[0] ? 0 : 32) | (amp->amp_vol[index][0] & 31); - vol[1] = (ucontrol->value.integer.value[1] ? 0 : 32) | (amp->amp_vol[index][1] & 31); + vol[0] = (ucontrol->value.integer.value[0] ? 0 : 32) + | (amp->amp_vol[index][0] & 31); + vol[1] = (ucontrol->value.integer.value[1] ? 0 : 32) + | (amp->amp_vol[index][1] & 31); return awacs_amp_set_vol(amp, index, vol[0], vol[1], 1); } @@ -403,7 +410,7 @@ static int snd_pmac_awacs_info_tone_amp( uinfo->value.integer.max = 14; return 0; } - + static int snd_pmac_awacs_get_tone_amp(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -445,7 +452,7 @@ static int snd_pmac_awacs_info_master_am uinfo->value.integer.max = 99; return 0; } - + static int snd_pmac_awacs_get_master_amp(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -493,14 +500,14 @@ static struct snd_kcontrol_new snd_pmac_ .private_value = AMP_CH_HD, }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Tone Control - Bass", + .name = "Tone Control - Bass Playback Volume", .info = snd_pmac_awacs_info_tone_amp, .get = snd_pmac_awacs_get_tone_amp, .put = snd_pmac_awacs_put_tone_amp, .private_value = 0, }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Tone Control - Treble", + .name = "Tone Control - Treble Playback Volume", .info = snd_pmac_awacs_info_tone_amp, .get = snd_pmac_awacs_get_tone_amp, .put = snd_pmac_awacs_put_tone_amp, @@ -544,7 +551,7 @@ static int snd_pmac_screamer_mic_boost_i uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; uinfo->value.integer.min = 0; - uinfo->value.integer.max = 2; + uinfo->value.integer.max = 3; return 0; } @@ -552,16 +559,14 @@ static int snd_pmac_screamer_mic_boost_g struct snd_ctl_elem_value *ucontrol) { struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); - int val; + int val = 0; unsigned long flags; spin_lock_irqsave(&chip->reg_lock, flags); if (chip->awacs_reg[6] & MASK_MIC_BOOST) - val = 2; - else if (chip->awacs_reg[0] & MASK_GAINLINE) - val = 1; - else - val = 0; + val |= 2; + if (chip->awacs_reg[0] & MASK_GAINLINE) + val |= 1; spin_unlock_irqrestore(&chip->reg_lock, flags); ucontrol->value.integer.value[0] = val; return 0; @@ -578,11 +583,10 @@ static int snd_pmac_screamer_mic_boost_p spin_lock_irqsave(&chip->reg_lock, flags); val0 = chip->awacs_reg[0] & ~MASK_GAINLINE; val6 = chip->awacs_reg[6] & ~MASK_MIC_BOOST; - if (ucontrol->value.integer.value[0] > 0) { + if (ucontrol->value.integer.value[0] & 1) val0 |= MASK_GAINLINE; - if (ucontrol->value.integer.value[0] > 1) - val6 |= MASK_MIC_BOOST; - } + if (ucontrol->value.integer.value[0] & 2) + val6 |= MASK_MIC_BOOST; if (val0 != chip->awacs_reg[0]) { snd_pmac_awacs_write_reg(chip, 0, val0); changed = 1; @@ -599,9 +603,32 @@ static int snd_pmac_screamer_mic_boost_p * lists of mixer elements */ static struct snd_kcontrol_new snd_pmac_awacs_mixers[] __initdata = { - AWACS_VOLUME("Master Playback Volume", 2, 6, 1), AWACS_SWITCH("Master Capture Switch", 1, SHIFT_LOOPTHRU, 0), - AWACS_VOLUME("Capture Volume", 0, 4, 0), + AWACS_VOLUME("Master Capture Volume", 0, 4, 0), +/* AWACS_SWITCH("Unknown Playback Switch", 6, SHIFT_PAROUT0, 0), */ +}; + +static struct snd_kcontrol_new snd_pmac_screamer_mixers_beige[] __initdata = { + AWACS_VOLUME("Master Playback Volume", 2, 6, 1), + AWACS_VOLUME("Play-through Playback Volume", 5, 6, 1), + AWACS_SWITCH("Line Capture Switch", 0, SHIFT_MUX_MIC, 0), + AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_LINE, 0), +}; + +static struct snd_kcontrol_new snd_pmac_screamer_mixers_imac[] __initdata = { + AWACS_VOLUME("Line out Playback Volume", 2, 6, 1), + AWACS_VOLUME("Master Playback Volume", 5, 6, 1), + AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0), +}; + +static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac7500[] __initdata = { + AWACS_VOLUME("Line out Playback Volume", 2, 6, 1), + AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0), + AWACS_SWITCH("Line Capture Switch", 0, SHIFT_MUX_MIC, 0), +}; + +static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac[] __initdata = { + AWACS_VOLUME("Master Playback Volume", 2, 6, 1), AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0), }; @@ -621,35 +648,61 @@ static struct snd_kcontrol_new snd_pmac_ static struct snd_kcontrol_new snd_pmac_awacs_master_sw __initdata = AWACS_SWITCH("Master Playback Switch", 1, SHIFT_HDMUTE, 1); +static struct snd_kcontrol_new snd_pmac_awacs_master_sw_imac __initdata = +AWACS_SWITCH("Line out Playback Switch", 1, SHIFT_HDMUTE, 1); + static struct snd_kcontrol_new snd_pmac_awacs_mic_boost[] __initdata = { - AWACS_SWITCH("Mic Boost", 0, SHIFT_GAINLINE, 0), + AWACS_SWITCH("Mic Boost Capture Switch", 0, SHIFT_GAINLINE, 0), }; static struct snd_kcontrol_new snd_pmac_screamer_mic_boost[] __initdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Mic Boost", + .name = "Mic Boost Capture Volume", .info = snd_pmac_screamer_mic_boost_info, .get = snd_pmac_screamer_mic_boost_get, .put = snd_pmac_screamer_mic_boost_put, }, }; +static struct snd_kcontrol_new snd_pmac_awacs_mic_boost_pmac7500[] __initdata = +{ + AWACS_SWITCH("Line Boost Capture Switch", 0, SHIFT_GAINLINE, 0), +}; + +static struct snd_kcontrol_new snd_pmac_screamer_mic_boost_beige[] __initdata = +{ + AWACS_SWITCH("Line Boost Capture Switch", 0, SHIFT_GAINLINE, 0), + AWACS_SWITCH("CD Boost Capture Switch", 6, SHIFT_MIC_BOOST, 0), +}; + +static struct snd_kcontrol_new snd_pmac_screamer_mic_boost_imac[] __initdata = +{ + AWACS_SWITCH("Line Boost Capture Switch", 0, SHIFT_GAINLINE, 0), + AWACS_SWITCH("Mic Boost Capture Switch", 6, SHIFT_MIC_BOOST, 0), +}; + static struct snd_kcontrol_new snd_pmac_awacs_speaker_vol[] __initdata = { AWACS_VOLUME("PC Speaker Playback Volume", 4, 6, 1), }; + static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw __initdata = AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_SPKMUTE, 1); +static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw_imac __initdata = +AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_PAROUT1, 0); + /* * add new mixer elements to the card */ -static int build_mixers(struct snd_pmac *chip, int nums, struct snd_kcontrol_new *mixers) +static int build_mixers(struct snd_pmac *chip, int nums, + struct snd_kcontrol_new *mixers) { int i, err; for (i = 0; i < nums; i++) { - if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&mixers[i], chip))) < 0) + if ((err = snd_ctl_add(chip->card, + snd_ctl_new1(&mixers[i], chip))) < 0) return err; } return 0; @@ -699,8 +752,10 @@ static void snd_pmac_awacs_resume(struct #ifdef PMAC_AMP_AVAIL if (chip->mixer_data) { struct awacs_amp *amp = chip->mixer_data; - awacs_amp_set_vol(amp, 0, amp->amp_vol[0][0], amp->amp_vol[0][1], 0); - awacs_amp_set_vol(amp, 1, amp->amp_vol[1][0], amp->amp_vol[1][1], 0); + awacs_amp_set_vol(amp, 0, + amp->amp_vol[0][0], amp->amp_vol[0][1], 0); + awacs_amp_set_vol(amp, 1, + amp->amp_vol[1][0], amp->amp_vol[1][1], 0); awacs_amp_set_tone(amp, amp->amp_tone[0], amp->amp_tone[1]); awacs_amp_set_master(amp, amp->amp_master); } @@ -708,6 +763,14 @@ static void snd_pmac_awacs_resume(struct } #endif /* CONFIG_PM */ +#define IS_PM7500 (machine_is_compatible("AAPL,7500")) +#define IS_BEIGE (machine_is_compatible("AAPL,Gossamer")) +#define IS_IMAC (machine_is_compatible("PowerMac2,1") \ + || machine_is_compatible("PowerMac2,2") \ + || machine_is_compatible("PowerMac4,1")) + +static int imac; + #ifdef PMAC_SUPPORT_AUTOMUTE /* * auto-mute stuffs @@ -750,9 +813,16 @@ static void snd_pmac_awacs_update_automu } else #endif { - int reg = chip->awacs_reg[1] | (MASK_HDMUTE|MASK_SPKMUTE); + int reg = chip->awacs_reg[1] + | (MASK_HDMUTE | MASK_SPKMUTE); + if (imac) { + reg &= ~MASK_SPKMUTE; + reg &= ~MASK_PAROUT1; + } if (snd_pmac_awacs_detect_headphone(chip)) reg &= ~MASK_HDMUTE; + else if (imac) + reg |= MASK_PAROUT1; else reg &= ~MASK_SPKMUTE; if (do_notify && reg == chip->awacs_reg[1]) @@ -778,8 +848,11 @@ static void snd_pmac_awacs_update_automu int __init snd_pmac_awacs_init(struct snd_pmac *chip) { + int pm7500 = IS_PM7500; + int beige = IS_BEIGE; int err, vol; + imac = IS_IMAC; /* looks like MASK_GAINLINE triggers something, so we set here * as start-up */ @@ -787,7 +860,7 @@ snd_pmac_awacs_init(struct snd_pmac *chi chip->awacs_reg[1] = MASK_CMUTE | MASK_AMUTE; /* FIXME: Only machines with external SRS module need MASK_PAROUT */ if (chip->has_iic || chip->device_id == 0x5 || - /*chip->_device_id == 0x8 || */ + /* chip->_device_id == 0x8 || */ chip->device_id == 0xb) chip->awacs_reg[1] |= MASK_PAROUT; /* get default volume from nvram */ @@ -798,8 +871,10 @@ snd_pmac_awacs_init(struct snd_pmac *chi chip->awacs_reg[2] = vol; chip->awacs_reg[4] = vol; if (chip->model == PMAC_SCREAMER) { - chip->awacs_reg[5] = vol; /* FIXME: screamer has loopthru vol control */ - chip->awacs_reg[6] = MASK_MIC_BOOST; /* FIXME: maybe should be vol << 3 for PCMCIA speaker */ + /* FIXME: screamer has loopthru vol control */ + chip->awacs_reg[5] = vol; + /* FIXME: maybe should be vol << 3 for PCMCIA speaker */ + chip->awacs_reg[6] = MASK_MIC_BOOST; chip->awacs_reg[7] = 0; } @@ -815,7 +890,8 @@ snd_pmac_awacs_init(struct snd_pmac *chi return -ENOMEM; chip->mixer_data = amp; chip->mixer_free = awacs_amp_free; - awacs_amp_set_vol(amp, 0, 63, 63, 0); /* mute and zero vol */ + /* mute and zero vol */ + awacs_amp_set_vol(amp, 0, 63, 63, 0); awacs_amp_set_vol(amp, 1, 63, 63, 0); awacs_amp_set_tone(amp, 7, 7); /* 0 dB */ awacs_amp_set_master(amp, 79); /* 0 dB */ @@ -826,20 +902,25 @@ snd_pmac_awacs_init(struct snd_pmac *chi /* set headphone-jack detection bit */ switch (chip->model) { case PMAC_AWACS: - chip->hp_stat_mask = 0x04; + chip->hp_stat_mask = pm7500 ? MASK_HDPCONN + : MASK_LOCONN; break; case PMAC_SCREAMER: switch (chip->device_id) { case 0x08: - /* 1 = side jack, 2 = front jack */ - chip->hp_stat_mask = 0x03; + case 0x0B: + chip->hp_stat_mask = imac + ? MASK_LOCONN_IMAC | + MASK_HDPLCONN_IMAC | + MASK_HDPRCONN_IMAC + : MASK_HDPCONN; break; case 0x00: case 0x05: - chip->hp_stat_mask = 0x04; + chip->hp_stat_mask = MASK_LOCONN; break; default: - chip->hp_stat_mask = 0x08; + chip->hp_stat_mask = MASK_HDPCONN; break; } break; @@ -857,15 +938,37 @@ snd_pmac_awacs_init(struct snd_pmac *chi if ((err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_mixers), snd_pmac_awacs_mixers)) < 0) return err; - if (chip->model == PMAC_SCREAMER) + if (beige) + ; + else if (chip->model == PMAC_SCREAMER) err = build_mixers(chip, ARRAY_SIZE(snd_pmac_screamer_mixers2), snd_pmac_screamer_mixers2); - else + else if (!pm7500) err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_mixers2), snd_pmac_awacs_mixers2); if (err < 0) return err; - chip->master_sw_ctl = snd_ctl_new1(&snd_pmac_awacs_master_sw, chip); + if (pm7500) + err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_awacs_mixers_pmac7500), + snd_pmac_awacs_mixers_pmac7500); + else if (beige) + err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_screamer_mixers_beige), + snd_pmac_screamer_mixers_beige); + else if (imac) + err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_screamer_mixers_imac), + snd_pmac_screamer_mixers_imac); + else + err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_awacs_mixers_pmac), + snd_pmac_awacs_mixers_pmac); + if (err < 0) + return err; + chip->master_sw_ctl = snd_ctl_new1((pm7500 || imac) + ? &snd_pmac_awacs_master_sw_imac + : &snd_pmac_awacs_master_sw, chip); if ((err = snd_ctl_add(chip->card, chip->master_sw_ctl)) < 0) return err; #ifdef PMAC_AMP_AVAIL @@ -876,35 +979,58 @@ snd_pmac_awacs_init(struct snd_pmac *chi * screamer registers. * in this case, it seems the route C is not used. */ - if ((err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_amp_vol), + if ((err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_awacs_amp_vol), snd_pmac_awacs_amp_vol)) < 0) return err; /* overwrite */ - chip->master_sw_ctl = snd_ctl_new1(&snd_pmac_awacs_amp_hp_sw, chip); + chip->master_sw_ctl = snd_ctl_new1(&snd_pmac_awacs_amp_hp_sw, + chip); if ((err = snd_ctl_add(chip->card, chip->master_sw_ctl)) < 0) return err; - chip->speaker_sw_ctl = snd_ctl_new1(&snd_pmac_awacs_amp_spk_sw, chip); + chip->speaker_sw_ctl = snd_ctl_new1(&snd_pmac_awacs_amp_spk_sw, + chip); if ((err = snd_ctl_add(chip->card, chip->speaker_sw_ctl)) < 0) return err; } else #endif /* PMAC_AMP_AVAIL */ { /* route A = headphone, route C = speaker */ - if ((err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_speaker_vol), + if ((err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_awacs_speaker_vol), snd_pmac_awacs_speaker_vol)) < 0) return err; - chip->speaker_sw_ctl = snd_ctl_new1(&snd_pmac_awacs_speaker_sw, chip); + chip->speaker_sw_ctl = snd_ctl_new1(imac + ? &snd_pmac_awacs_speaker_sw_imac + : &snd_pmac_awacs_speaker_sw, chip); if ((err = snd_ctl_add(chip->card, chip->speaker_sw_ctl)) < 0) return err; } - if (chip->model == PMAC_SCREAMER) { - if ((err = build_mixers(chip, ARRAY_SIZE(snd_pmac_screamer_mic_boost), - snd_pmac_screamer_mic_boost)) < 0) + if (beige) { + if ((err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_screamer_mic_boost_beige), + snd_pmac_screamer_mic_boost_beige)) < 0) + return err; + } else if (imac) { + if ((err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_screamer_mic_boost_imac), + snd_pmac_screamer_mic_boost_imac)) < 0) + return err; + } else if (chip->model == PMAC_SCREAMER) { + if ((err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_screamer_mic_boost), + snd_pmac_screamer_mic_boost)) < 0) + return err; + } else if (pm7500) { + if ((err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_awacs_mic_boost_pmac7500), + snd_pmac_awacs_mic_boost_pmac7500)) < 0) return err; } else { - if ((err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_mic_boost), - snd_pmac_awacs_mic_boost)) < 0) + if ((err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_awacs_mic_boost), + snd_pmac_awacs_mic_boost)) < 0) return err; } --- sound/ppc/awacs.h.orig 2008-04-05 17:45:08.000000000 +0300 +++ sound/ppc/awacs.h 2008-04-07 11:26:58.198763416 +0300 @@ -116,6 +116,11 @@ struct awacs_regs { #define MASK_HDMUTE MASK_AMUTE #define SHIFT_HDMUTE 9 #define MASK_PAROUT (0x3 << 10) /* Parallel Out (???) */ +#define MASK_PAROUT0 (0x1 << 10) /* Parallel Out (???) */ +#define MASK_PAROUT1 (0x1 << 11) /* Parallel Out (enable speaker) */ +#define SHIFT_PAROUT 10 +#define SHIFT_PAROUT0 10 +#define SHIFT_PAROUT1 11 #define SAMPLERATE_48000 (0x0 << 3) /* 48 or 44.1 kHz */ #define SAMPLERATE_32000 (0x1 << 3) /* 32 or 29.4 kHz */ @@ -152,8 +157,15 @@ struct awacs_regs { #define MASK_REVISION (0xf << 12) /* Revision Number */ #define MASK_MFGID (0xf << 8) /* Mfg. ID */ #define MASK_CODSTATRES (0xf << 4) /* bits 4 - 7 reserved */ -#define MASK_INPPORT (0xf) /* Input Port */ -#define MASK_HDPCONN 8 /* headphone plugged in */ +#define MASK_INSENSE (0xf) /* port sense bits: */ +#define MASK_HDPCONN 8 /* headphone plugged in */ +#define MASK_LOCONN 4 /* line-out plugged in */ +#define MASK_LICONN 2 /* line-in plugged in */ +#define MASK_MICCONN 1 /* microphone plugged in */ +#define MASK_LICONN_IMAC 8 /* line-in plugged in */ +#define MASK_HDPRCONN_IMAC 4 /* headphone right plugged in */ +#define MASK_HDPLCONN_IMAC 2 /* headphone left plugged in */ +#define MASK_LOCONN_IMAC 1 /* line-out plugged in */ /* Clipping Count Reg Bit Masks */ /* -------- ----- --- --- ----- */ @@ -163,7 +175,8 @@ struct awacs_regs { /* DBDMA ChannelStatus Bit Masks */ /* ----- ------------- --- ----- */ #define MASK_CSERR (0x1 << 7) /* Error */ -#define MASK_EOI (0x1 << 6) /* End of Input -- only for Input Channel */ +#define MASK_EOI (0x1 << 6) /* End of Input -- + only for Input Channel */ #define MASK_CSUNUSED (0x1f << 1) /* bits 1-5 not used */ #define MASK_WAIT (0x1) /* Wait */

2 3

[alsa-devel] [PATCH 001b/003] snd-powermac: style pmac.c
by Risto Suominen 16 Apr '08

16 Apr '08

From: Risto Suominen <Risto.Suominen(a)gmail.com> Coding style corrections for pmac.c. Signed-off-by: Risto Suominen <Risto.Suominen(a)gmail.com> --- Kernel version 2.6.25-rc8 (probably any 2.6) --- sound/ppc/pmac.c.orig 2008-04-05 17:45:08.000000000 +0300 +++ sound/ppc/pmac.c 2008-04-07 12:18:47.503077312 +0300 @@ -214,7 +214,7 @@ static int snd_pmac_pcm_prepare(struct s int rate_index; long offset; struct pmac_stream *astr; - + rec->dma_size = snd_pcm_lib_buffer_bytes(subs); rec->period_size = snd_pcm_lib_period_bytes(subs); rec->nperiods = rec->dma_size / rec->period_size; @@ -643,7 +643,7 @@ static int snd_pmac_pcm_close(struct snd /* reset constraints */ astr->cur_freqs = chip->freqs_ok; astr->cur_formats = chip->formats_ok; - + return 0; }

1 0

[alsa-devel] [PATCH 001a/003] snd-powermac: enable headphone detection
by Risto Suominen 16 Apr '08

16 Apr '08

From: Risto Suominen <Risto.Suominen(a)gmail.com> Enable port change interrupt while initialising AWACS, Screamer, and Burgundy chipsets. Signed-off-by: Risto Suominen <Risto.Suominen(a)gmail.com> --- Kernel version 2.6.25-rc8 (probably any 2.6) --- sound/ppc/pmac.c.orig 2008-04-05 17:45:08.000000000 +0300 +++ sound/ppc/pmac.c 2008-04-07 12:18:47.503077312 +0300 @@ -1300,9 +1304,9 @@ int __init snd_pmac_new(struct snd_card snd_pmac_sound_feature(chip, 1); - /* reset */ - if (chip->model == PMAC_AWACS) - out_le32(&chip->awacs->control, 0x11); + /* reset & enable interrupts */ + if (chip->model <= PMAC_BURGUNDY) + out_le32(&chip->awacs->control, chip->control_mask); /* Powerbooks have odd ways of enabling inputs such as an expansion-bay CD or sound from an internal modem

1 0

[alsa-devel] [Fwd: Another unknown bt878]
by John R. 16 Apr '08

16 Apr '08

This card is sold as the PV-981. I tried various digital_rate settings. -------- Original Message -------- Subject: Another unknown bt878 Date: Mon, 14 Apr 2008 23:17:47 -0500 From: John R. <johnr(a)wowway.com> To: alsa-devel(a)alsa-project.org Sound doesn't work on my bt878 based card. Let me know what other information is needed... kernel 2.6.24.2: Apr 14 20:28:38 server kernel: unknown card 0x878-0xaa00:0x1460 Apr 14 20:28:38 server kernel: please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)alsa-project.org> Apr 14 20:28:38 server kernel: ACPI: PCI Interrupt 0000:05:00.1[A] -> GSI 18 (level, low) -> IRQ 18 Apr 14 20:28:38 server kernel: bt87x0: Using board 0, analog, digital (rate 32000 Hz) Apr 14 20:28:38 server kernel: unknown card 0x878-0xaa01:0x1461 Apr 14 20:28:38 server kernel: please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)alsa-project.org> Apr 14 20:28:38 server kernel: ACPI: PCI Interrupt 0000:05:01.1[A] -> GSI 19 (level, low) -> IRQ 19 Apr 14 20:28:38 server kernel: bt87x1: Using board 0, analog, digital (rate 32000 Hz) Apr 14 20:28:38 server kernel: unknown card 0x878-0xaa02:0x1462 Apr 14 20:28:38 server kernel: please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)alsa-project.org> Apr 14 20:28:38 server kernel: ACPI: PCI Interrupt 0000:05:02.1[A] -> GSI 16 (level, low) -> IRQ 16 Apr 14 20:28:38 server kernel: bt87x2: Using board 0, analog, digital (rate 3200 0 Hz) Apr 14 20:28:38 server kernel: unknown card 0x878-0xaa03:0x1463 Apr 14 20:28:38 server kernel: please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)alsa-project.org> Apr 14 20:28:38 server kernel: ACPI: PCI Interrupt 0000:05:03.1[A] -> GSI 17 (level, low) -> IRQ 17 Apr 14 20:28:38 server kernel: bt87x3: Using board 0, analog, digital (rate 32000 Hz) lspci: 05:00.0 Multimedia video controller: Brooktree Corporation Bt878 Video Capture (rev 11) 05:00.1 Multimedia controller: Brooktree Corporation Bt878 Audio Capture (rev 11) 05:01.0 Multimedia video controller: Brooktree Corporation Bt878 Video Capture (rev 11) 05:01.1 Multimedia controller: Brooktree Corporation Bt878 Audio Capture (rev 11) 05:02.0 Multimedia video controller: Brooktree Corporation Bt878 Video Capture (rev 11) 05:02.1 Multimedia controller: Brooktree Corporation Bt878 Audio Capture (rev 11) 05:03.0 Multimedia video controller: Brooktree Corporation Bt878 Video Capture (rev 11) 05:03.1 Multimedia controller: Brooktree Corporation Bt878 Audio Capture (rev 11) at boot: bttv: driver version 0.9.17 loaded bttv: using 8 buffers with 2080k (520 pages) each for capture bttv: Bt8xx card found (0). ACPI: PCI Interrupt 0000:05:00.0[A] -> GSI 18 (level, low) -> IRQ 18 bttv0: Bt878 (rev 17) at 0000:05:00.0, irq: 18, latency: 64, mmio: 0xdfcfe000 bttv0: detected: Provideo PV150A-1 [card=98], PCI subsystem ID is aa00:1460 bttv0: using: ProVideo PV150 [card=98,autodetected] bttv0: gpio: en=00000000, out=00000000 in=00ffffff [init] bttv0: tuner absent bttv0: registered device video0 bttv0: registered device vbi0 bttv0: PLL: 28636363 => 35468950 .. ok bttv: Bt8xx card found (1). ACPI: PCI Interrupt 0000:05:01.0[A] -> GSI 19 (level, low) -> IRQ 19 bttv1: Bt878 (rev 17) at 0000:05:01.0, irq: 19, latency: 64, mmio: 0xdfcfc000 bttv1: detected: Provideo PV150A-2 [card=98], PCI subsystem ID is aa01:1461 bttv1: using: ProVideo PV150 [card=98,autodetected] bttv1: gpio: en=00000000, out=00000000 in=00ffffff [init] bttv1: tuner absent bttv1: registered device video1 bttv1: registered device vbi1 bttv1: PLL: 28636363 => 35468950 .. ok bttv: Bt8xx card found (2). ACPI: PCI Interrupt 0000:05:02.0[A] -> GSI 16 (level, low) -> IRQ 16 bttv2: Bt878 (rev 17) at 0000:05:02.0, irq: 16, latency: 64, mmio: 0xdfcfa000 bttv2: detected: Provideo PV150A-3 [card=98], PCI subsystem ID is aa02:1462 bttv2: using: ProVideo PV150 [card=98,autodetected] bttv2: gpio: en=00000000, out=00000000 in=00ffffff [init] bttv2: tuner absent bttv2: registered device video2 bttv2: registered device vbi2 bttv2: PLL: 28636363 => 35468950 .. ok bttv: Bt8xx card found (3). ACPI: PCI Interrupt 0000:05:03.0[A] -> GSI 17 (level, low) -> IRQ 17 bttv3: Bt878 (rev 17) at 0000:05:03.0, irq: 17, latency: 64, mmio: 0xdfcf8000 bttv3: detected: Provideo PV150A-4 [card=98], PCI subsystem ID is aa03:1463 bttv3: using: ProVideo PV150 [card=98,autodetected] bttv3: gpio: en=00000000, out=00000000 in=00ffffff [init] bttv3: tuner absent bttv3: registered device video3 bttv3: registered device vbi3 bttv3: PLL: 28636363 => 35468950 .. ok

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[alsa-devel] snd_ctl_elem_write fails -EPERM if snd_pcm is open
by Bouterse, Bob 15 Apr '08

15 Apr '08

I am trying to sent IEC958 status bits on a VT1708A HAD-INTEL codec while audio is being output. snd_ctl_elem_write returns -EPERM from linux-2.6.24.3/sound/core/control.c:769 . Is there some way I can do this? FYI using iecset will also give you an: # iecset copywrite on snd_ctl_elem_write: Operation not permitted I am using alsa-lib-1.0.14a $uname -a Linux vision-vs 2.6.24.3 #1 SMP Fri Mar 28 09:35:45 EDT 2008 i686 unknown $ gcc -v Target: x86_64-linux-gnu Configured with: ../src/configure -v --enable-languages=c,c++,fortran,objc,obj-c++,treelang --prefix=/usr --enable-shared --with-system-zlib --libexecdir=/usr/lib --without-included-gettext --enable-threads=posix --enable-nls --with-gxx-include-dir=/usr/include/c++/4.2 --program-suffix=-4.2 --enable-clocale=gnu --enable-libstdcxx-debug --enable-mpfr --enable-checking=release --build=x86_64-linux-gnu --host=x86_64-linux-gnu --target=x86_64-linux-gnu Thread model: posix gcc version 4.2.3 20071123 (prerelease) (Debian 4.2.2-4) If it helps, here is the code I am using to set the bits. I modeled this after the iecset.c code. Everything opens fine, it is the final call to "snd_ctl_elem_write(this->ctl_handle, cval))" that fails. static int ao_alsa_set_spdif_status(ao_driver_t *this_gen, snd_aes_iec958_t* iec958) { alsa_driver_t *this = (alsa_driver_t *) this_gen; config_values_t *config = this->class->xine->config; char *pcm_device; snd_ctl_card_info_t *hw_info; int err; snd_pcm_info_t *pcm_info; int card; char card_name[20]; snd_ctl_elem_list_t *clist; snd_ctl_elem_id_t *cid; snd_ctl_elem_value_t *cval; unsigned int controls, cidx; const char *spdif_str = SND_CTL_NAME_IEC958("", PLAYBACK, DEFAULT); int opened_audio=0; llprintf(LOG_TRACE_ENTER, "Entered\n"); if(!this->ctl_handle) { if (!this->audio_fd) { pcm_device = config->lookup_entry(config, "audio.device.alsa_passthrough_device")->str_value; llprintf(LOG_MIXER,"Calling local_snd_pcm_open(%s)\n", pcm_device); err=local_snd_pcm_open(&this->audio_fd, pcm_device, SND_PCM_STREAM_PLAYBACK, 1); /* NON-BLOCK mode */ if ( err <0 ) { xine_log (this->class->xine, XINE_LOG_MSG, _("local_snd_pcm_open() failed:%d:%s\n"), err, snd_strerror(err)); return -1; } opened_audio = 1; } snd_pcm_info_alloca(&pcm_info); snd_pcm_info(this->audio_fd, pcm_info); card = snd_pcm_info_get_card(pcm_info); llprintf(LOG_MIXER,"snd_pcm_info_get_card() returned %d\n", card); if(card < 0){ sprintf(card_name, "default"); } else{ sprintf(card_name, "hw:CARD=%i", card); } if (opened_audio && this->audio_fd) { snd_pcm_close(this->audio_fd); this->audio_fd = NULL; opened_audio = 0; } llprintf(LOG_MIXER,"Calling snd_ctl_open(%s)\n", card_name); if ( (err = snd_ctl_open (&this->ctl_handle, card_name, 0)) < 0 ) { xprintf (this->class->xine, XINE_VERBOSITY_DEBUG, "%s: snd_ctl_open(): %s\n", LOG_MODULE, snd_strerror(err)); return -1; } } snd_ctl_card_info_alloca(&hw_info); if ( (err = snd_ctl_card_info (this->ctl_handle, hw_info)) < 0 ) { xprintf (this->class->xine, XINE_VERBOSITY_DEBUG, "%s: snd_ctl_card_info(): %s\n", LOG_MODULE, snd_strerror(err)); snd_ctl_close(this->ctl_handle); this->ctl_handle = 0; return -1; } /* set the spdif status bits */ snd_ctl_elem_list_alloca(&clist); if ((err = snd_ctl_elem_list(this->ctl_handle, clist)) < 0) { xprintf (this->class->xine, XINE_VERBOSITY_DEBUG, "snd_ctl_elem_list(): %s\n", snd_strerror(err)); snd_ctl_close(this->ctl_handle); this->ctl_handle = 0; return -1; } if ((err = snd_ctl_elem_list_alloc_space(clist, snd_ctl_elem_list_get_count(clist))) < 0) { xprintf (this->class->xine, XINE_VERBOSITY_DEBUG, "snd_ctl_elem_list_alloc_space(): %s\n", snd_strerror(err)); snd_ctl_close(this->ctl_handle); return -1; } if ((err = snd_ctl_elem_list(this->ctl_handle, clist)) < 0) { xprintf (this->class->xine, XINE_VERBOSITY_DEBUG, "snd_ctl_elem_list(): %s\n", snd_strerror(err)); snd_ctl_close(this->ctl_handle); this->ctl_handle = 0; return -1; } controls = snd_ctl_elem_list_get_used(clist); for (cidx = 0; cidx < controls; cidx++) { if (!strcmp(snd_ctl_elem_list_get_name(clist, cidx), spdif_str)) break; } if (cidx >= controls) { fprintf(stderr, "control \"%s\" not found\n", spdif_str); snd_ctl_close(this->ctl_handle); this->ctl_handle = 0; return -1; } snd_ctl_elem_id_alloca(&cid); snd_ctl_elem_list_get_id(clist, cidx, cid); snd_ctl_elem_value_alloca(&cval); snd_ctl_elem_value_set_id(cval, cid); llprintf(LOG_MIXER,"Writing AES0=0x%02x,AES1=0x%02x,AES2=0x%02x,AES3=0x%02x\n", iec958->status[0], iec958->status[1], iec958->status[2], iec958->status[3]); /* store the values */ snd_ctl_elem_value_set_iec958(cval, iec958); if ((err = snd_ctl_elem_write(this->ctl_handle, cval)) < 0) { xprintf (this->class->xine, XINE_VERBOSITY_DEBUG, "snd_ctl_elem_write(): %s\n", snd_strerror(err)); snd_ctl_close(this->ctl_handle); this->ctl_handle = 0; } llprintf(LOG_MIXER,"snd_ctl_elem_write() returrned %d\n",err); /*-------------------------------------------------------*/ /*snd_ctl_close (this->ctl_handle);*/ return 0; } ---------------------------------------------------- Bob Bouterse Sr. Software Engineer D&M Holdings Inc.

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[alsa-devel] [PATCH 001/003] snd-powermac: enable headphone detection
by Risto Suominen 15 Apr '08

15 Apr '08

From: Risto Suominen <Risto.Suominen at gmail.com> Enable port change interrupt while initialising AWACS, Screamer, and Burgundy chipsets. Signed-off-by: Risto Suominen <Risto.Suominen at gmail.com> --- Kernel version 2.6.25-rc8 (probably any 2.6) --- sound/ppc/pmac.c.orig 2008-04-05 17:45:08.000000000 +0300 +++ sound/ppc/pmac.c 2008-04-07 12:18:47.503077312 +0300 @@ -214,7 +214,7 @@ static int snd_pmac_pcm_prepare(struct s int rate_index; long offset; struct pmac_stream *astr; - + rec->dma_size = snd_pcm_lib_buffer_bytes(subs); rec->period_size = snd_pcm_lib_period_bytes(subs); rec->nperiods = rec->dma_size / rec->period_size; @@ -643,7 +643,7 @@ static int snd_pmac_pcm_close(struct snd /* reset constraints */ astr->cur_freqs = chip->freqs_ok; astr->cur_formats = chip->formats_ok; - + return 0; } @@ -1063,6 +1063,10 @@ static int __init snd_pmac_detect(struct else { struct pci_dev *pdev = NULL; +#ifndef for_each_pci_dev +#define for_each_pci_dev(d) \ + while ((d = pci_get_device(PCI_ANY_ID, PCI_ANY_ID, d)) != NULL) +#endif for_each_pci_dev(pdev) { struct device_node *np = pci_device_to_OF_node(pdev); if (np && np == macio->of_node) { @@ -1300,9 +1304,9 @@ int __init snd_pmac_new(struct snd_card snd_pmac_sound_feature(chip, 1); - /* reset */ - if (chip->model == PMAC_AWACS) - out_le32(&chip->awacs->control, 0x11); + /* reset & enable interrupts */ + if (chip->model <= PMAC_BURGUNDY) + out_le32(&chip->awacs->control, chip->control_mask); /* Powerbooks have odd ways of enabling inputs such as an expansion-bay CD or sound from an internal modem

2 2

[alsa-devel] ac97_bus: exports duplicate symbol ac97_bus_type (owned by kernel)
by Vitaly V. Ch 15 Apr '08

15 Apr '08

#modprobe snd_via82xx #dmesg|tail ac97_bus: exports duplicate symbol ac97_bus_type (owned by kernel) snd_ac97_codec: disagrees about version of symbol ac97_bus_type snd_ac97_codec: Unknown symbol ac97_bus_type snd_via82xx: Unknown symbol snd_ac97_resume snd_via82xx: Unknown symbol snd_ac97_set_rate snd_via82xx: Unknown symbol snd_ac97_update_bits snd_via82xx: Unknown symbol snd_ac97_mixer snd_via82xx: Unknown symbol snd_ac97_bus snd_via82xx: Unknown symbol snd_ac97_update_power snd_via82xx: Unknown symbol snd_ac97_suspend snd_via82xx: Unknown symbol snd_ac97_get_short_name snd_via82xx: Unknown symbol snd_ac97_tune_hardware any ideas? \\wbr Vitaly

1 0

[alsa-devel] PXA AC97 updates
by Mark Brown 15 Apr '08

15 Apr '08

This patch series fixes a bug in the currently queued conversion of the PXA AC97 drivers to the clock API and adds support for AC97 on PXA3xx SoCs. Takashi, since these changes build on things currently queued in the ARM tree for 2.6.26, including ARM-specific changes, it's probably best to merge via there if that's OK?

3 10

Re: [alsa-devel] no front speaker sound with ALC262
by Rene Dohmen 15 Apr '08

15 Apr '08

Tobin, Did you receive the unit? I'm wondering if you had some progress already. Greets, Rene Op Mar 31, 2008, om 3:30 PM heeft Tobin Davis het volgende geschreven: > Yes, it would. Hmmm > > Well, I hope they send the unit today. > > Tobin > > On Mon, 2008-03-31 at 13:57 +0200, Rene Dohmen wrote: >> >> with the preinstalled win XP the sound is working OK. >> So that would rule out hardware problems with the soundcard. >> >> >> Rene >> >> >> >> >> >> >> Op Mar 28, 2008, om 7:28 PM heeft Tobin Davis het volgende >> geschreven: >> >>> I'm still waiting for my demo unit from Tangent. Latest update >>> is they think it may be a hardware issue (missing a resistor). I >>> should know more Monday. If this is the case, then the driver >>> should work with very minor modifications (subsystem ID addition). >>> >>> Tobin >>> >>> >>> On Tue, 2008-03-25 at 11:40 +0100, Rene Dohmen wrote: >>>> >>>> Op Mar 22, 2008, om 8:09 PM heeft Tobin Davis het volgende >>>> geschreven: >>>> >>>>> Rene, don't bother bugging Tangent. I have a demo unit on the >>>>> way. My first priority is getting the driver working. >>>>> >>>>> Tangent really won't help much - their a distributor, not the >>>>> OEM. This is an MSI Crystal 945 in a Tangent shell. >>>>> >>>> >>>> >>>> I know that the system is actually a Crystal 945. I got the >>>> touchscreen working via a dev driver from salt (so not tangent) >>>> On the mainboard there is a conenctor: JFP1that goes to the >>>> ms4088 board. The mainboard connects to: >>>> - power led >>>> - hdd led >>>> - speakers >>>> - main power switch >>>> >>>> >>>> I can provide you with some photoos if you like. >>>> >>>> >>>> As stated earlier: the sound is working by with very low volume >>>> levels >>>> >>>> >>>> >>>> >>>>> If you want to try something, see if the bios can be reflashed >>>>> with the Crystal 945 firmware. >>>>> >>>> >>>> >>>> I sound working with the original crystal? I don't think tangent >>>> has flshed their BIOS to another version. >>>> In the crystal 945 specs I found that the're using a Realtek >>>> ALC262 with azalia codec, maybe this helps. >>>> >>>> >>>> Rene >>>> >>>> >>>> >>>>> As to the speaker board, where on the main board is it >>>>> mounted? Is the header labeled? How many pins? It sounds >>>>> like the speakers may be on the front panel header, which would >>>>> make them function like headphones. Or this board is custom >>>>> build for this application and I don't know what I'm talking >>>>> about. >>>>> >>>>> Tobin >>>>> >>>>> >>>>> On Sat, 2008-03-22 at 18:59 +0100, Rene Dohmen wrote: >>>>>> >>>>>> Hi Guys, >>>>>> >>>>>> I'm glad t o read that there are other people with the same >>>>>> problem >>>>>> on the same system. >>>>>> I just opened my Tangent 8000S and it has indeed a MSI mainboard: >>>>>> MS7290. The speakers are connected to a small custom board: >>>>>> MS-4088 >>>>>> There's no extra chip on this customboard and the board is >>>>>> directly >>>>>> connected to the mainboard. So probably the signal leaving the >>>>>> mainboard in amplified already. (or there's a amplifier >>>>>> builtin the >>>>>> speakers) >>>>>> >>>>>> Next week i'll have some time to put tangent under some more >>>>>> pressure >>>>>> for info, so maybe that will speed up things a little bit. >>>>>> >>>>>> Grz Rene >>>>>> >>>>>> _______________________________________________ >>>>>> Alsa-devel mailing list >>>>>> Alsa-devel(a)alsa-project.org >>>>>> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >>>>> -- Tobin Davis >>>>> >>>>> love, n.: >>>>> When you don't want someone too close--because you're very >>>>> sensitive >>>>> to pleasure. >>>> >>> -- Tobin Davis >>> >>> I can hire one half of the working class to kill the other half. >>> -- Jay Gould >> >> > -- Tobin Davis > > He is no lawyer who cannot take two sides.

2 1

[alsa-devel] dB, dB and more dB....
by Tom Watson 15 Apr '08

15 Apr '08

I find all of this an interesting discussion, and it looks like: 1) People want to hear some output. 2) "Pro" people don't want to blow up speakers. <soapbox> As was said before it is all relative. I wrote an application using ALSA and also needed to allow it to play back CDs at the same level. What I did was take a sample WAV file and burn it to a CD. It had several levels, from full scale, and 3dB (divide by 2) down. It was quite informative for setting mixer levels. I needed it because, while I was using an "add-in" board (M-Audio Revo 7.1), the CD drive was connected to the internal motherboard input place. I used a small jumper plug to connect the output of the motherboard audio out to the Revo7.1 board "in" and setup mixers accordingly. I ended up using a scope to get the levels correct, without overloading (distorting) the audio path. Quite informative. It seems that all of us have differing ideas on "what to expect" when it comes to audio interfaces. Twenty years ago, these problems really didn't exist, since computer audio was in its infancy, but now audio is everywhere, and it muct be accounted for. Alsa's implementation of having audio MUTED on initialization (startup) is a good one, and many distributions have init scripts that take over from there, restoring old values. While not ideal (just set the audio to where I want it!), it is about the best we can do. The "pro" guy who doesn't want to blow up his speakers, I can only offer a suggestion. What do you do when plugging in and out "normal" audio equipment. Most of the time I turn down the level control since my big fat fingers will end up touching the tip/ring of the audio connector and LOTS of hum results. If you don't do this, you may have a PEBKAC problem which I (or any other developer) can solve. I think we are all chasing a problem that really doesn't exist. As for dB, some people refer to 0dB as "full scale". Some refer to it as 0 VU. Some make it out to be 1-v p-p. Go figure. The biggest problem is that crazy people think "louder" is better. It isn't! A small "white paper" on the subject (part of ALSA?) with an agreed upon "introduction to computer audio" might be a good place to start. Hopefully no more than a couple of pages in terms that everyone can understand. </soapbox> -- Tom Watson tsw(a)johana.com ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

2 1

[alsa-devel] [PATCH] ASoC: Add support for 19.2 MHz MCLK in TLV320AIC3X
by Jarkko Nikula 15 Apr '08

15 Apr '08

Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- sound/soc/codecs/tlv320aic3x.c | 11 +++++++++++ 1 files changed, 11 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index e6a730b..630684f 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -661,42 +661,52 @@ struct aic3x_rate_divs { static const struct aic3x_rate_divs aic3x_divs[] = { /* 8k */ {12000000, 8000, 48000, 0xa, 16, 3840}, + {19200000, 8000, 48000, 0xa, 10, 2400}, {22579200, 8000, 48000, 0xa, 8, 7075}, {33868800, 8000, 48000, 0xa, 5, 8049}, /* 11.025k */ {12000000, 11025, 44100, 0x6, 15, 528}, + {19200000, 11025, 44100, 0x6, 9, 4080}, {22579200, 11025, 44100, 0x6, 8, 0}, {33868800, 11025, 44100, 0x6, 5, 3333}, /* 16k */ {12000000, 16000, 48000, 0x4, 16, 3840}, + {19200000, 16000, 48000, 0x4, 10, 2400}, {22579200, 16000, 48000, 0x4, 8, 7075}, {33868800, 16000, 48000, 0x4, 5, 8049}, /* 22.05k */ {12000000, 22050, 44100, 0x2, 15, 528}, + {19200000, 22050, 44100, 0x2, 9, 4080}, {22579200, 22050, 44100, 0x2, 8, 0}, {33868800, 22050, 44100, 0x2, 5, 3333}, /* 32k */ {12000000, 32000, 48000, 0x1, 16, 3840}, + {19200000, 32000, 48000, 0x1, 10, 2400}, {22579200, 32000, 48000, 0x1, 8, 7075}, {33868800, 32000, 48000, 0x1, 5, 8049}, /* 44.1k */ {12000000, 44100, 44100, 0x0, 15, 528}, + {19200000, 44100, 44100, 0x0, 9, 4080}, {22579200, 44100, 44100, 0x0, 8, 0}, {33868800, 44100, 44100, 0x0, 5, 3333}, /* 48k */ {12000000, 48000, 48000, 0x0, 16, 3840}, + {19200000, 48000, 48000, 0x0, 10, 2400}, {22579200, 48000, 48000, 0x0, 8, 7075}, {33868800, 48000, 48000, 0x0, 5, 8049}, /* 64k */ {12000000, 64000, 96000, 0x1, 16, 3840}, + {19200000, 64000, 96000, 0x1, 10, 2400}, {22579200, 64000, 96000, 0x1, 8, 7075}, {33868800, 64000, 96000, 0x1, 5, 8049}, /* 88.2k */ {12000000, 88200, 88200, 0x0, 15, 528}, + {19200000, 88200, 88200, 0x0, 9, 4080}, {22579200, 88200, 88200, 0x0, 8, 0}, {33868800, 88200, 88200, 0x0, 5, 3333}, /* 96k */ {12000000, 96000, 96000, 0x0, 16, 3840}, + {19200000, 96000, 96000, 0x0, 10, 2400}, {22579200, 96000, 96000, 0x0, 8, 7075}, {33868800, 96000, 96000, 0x0, 5, 8049}, }; @@ -818,6 +828,7 @@ static int aic3x_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, switch (freq) { case 12000000: + case 19200000: case 22579200: case 33868800: aic3x->sysclk = freq; -- 1.5.4.4

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[alsa-devel] Patchmix-style application for EMU cards
by Camilo Polymeris 15 Apr '08

15 Apr '08

Hello I am working on a graphical mixing/routing application for the Emu10k series of semi-pro soundcards, but I am having trouble figuring out the routing mechanism of the driver, specially the "Multichannel PCM" part. Is there any technical documentation on this card I could read (I haven't found any), or maybe the driver developer can help me out? Thanks, Camilo.

1 0

[alsa-devel] line6usb in alsa
by Aldrin Martoq 15 Apr '08

15 Apr '08

There is a "driver" which I long wanted to be in official alsa: line6usb. http://www.tanzband-scream.at/line6/ While the code exists, I looked at it and in my opinion it's not ready for inclusion in alsa. For example, there is some MIDI code which belongs to userspace but it's in the kernel module. I have already contacted the maintainer of the project, so we can make this happen. My question are: - what "communication" steps do we need to put this code here? - do we have to put the initial line6usb support in one single patch or a step by step is preferred? - once the code is here, how can we improve it or make changes? - any other trap we could avoid right now? any suggestion of how can we make this the smoothest way possible? Thanks a lot! -- Aldrin Martoq Episodio 002 (Lunes 24 Marzo)! http://aldrinvideopodcast.podshow.com/

1 0

[alsa-devel] via82xx: how to get microphone working?
by Németh Márton 14 Apr '08

14 Apr '08

Hi, I have a Clevo D410J laptop with the following integrated sound card: 00:11.5 Multimedia audio controller: VIA Technologies, Inc. VT8233/A/8235/8237 AC97 Audio Controller (rev 50) Subsystem: CLEVO/KAPOK Computer Unknown device 4702 Flags: medium devsel, IRQ 20 I/O ports at 1400 [size=256] Capabilities: [c0] Power Management version 2 Kernel driver in use: VIA 82xx Audio Kernel modules: snd-via82xx There is also an integrated modem which also appears in kmix/alsamixer: 00:11.6 Communication controller: VIA Technologies, Inc. AC'97 Modem Controller (rev 80) Subsystem: CLEVO/KAPOK Computer Unknown device 4702 Flags: medium devsel, IRQ 20 I/O ports at 1800 [size=256] Capabilities: [d0] Power Management version 2 Kernel driver in use: VIA 82xx Modem Kernel modules: snd-via82xx-modem I am using Linux 2.6.23. I enabled the microphone input, increased the microphone record level to maximum, and enabled "Mic Boost". I can hear my voice through the speakers, but I cannot record sound (for example Skype cannot record my sound). The playback is working fine without any problem. Is there any setting I miss? $ lsmod |grep snd snd_via82xx 22572 1 snd_mpu401_uart 7064 1 snd_via82xx snd_seq_oss 29328 0 snd_seq_midi 6432 0 snd_seq_midi_event 6536 2 snd_seq_oss,snd_seq_midi snd_via82xx_modem 11408 1 snd_ac97_codec 95676 2 snd_via82xx,snd_via82xx_modem ac97_bus 2432 1 snd_ac97_codec snd_pcm_oss 38208 0 snd_mixer_oss 14216 1 snd_pcm_oss snd_seq 45752 5 snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_pcm 69420 4 snd_via82xx,snd_via82xx_modem,snd_ac97_codec,snd_pcm_oss snd_timer 19356 2 snd_seq,snd_pcm snd_rawmidi 19136 2 snd_mpu401_uart,snd_seq_midi snd_seq_device 6932 4 snd_seq_oss,snd_seq_midi,snd_seq,snd_rawmidi snd 45476 16 snd_via82xx,snd_mpu401_uart,snd_seq_oss,snd_via82xx_modem,snd_ac97_codec,snd_pcm_oss,snd_mixer_oss,snd_seq,snd_pcm,snd_timer,snd_rawmidi,snd_seq_device snd_page_alloc 8072 3 snd_via82xx,snd_via82xx_modem,snd_pcm soundcore 6496 1 snd Regards, Márton Németh

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[alsa-devel] [PATCH 003/003] snd-powermac: Burgundy mixers for B&W and iMac
by Risto Suominen 14 Apr '08

14 Apr '08

From: Risto Suominen <Risto.Suominen at gmail.com> Add mixer controls and correct headphone detection bits for PowerMac G3 B&W and iMac G3 Tray-loading, both having Burgundy chipset. Signed-off-by: Risto Suominen <Risto.Suominen at gmail.com> --- Kernel version 2.6.25-rc8 (probably any 2.6) --- sound/ppc/burgundy.c.orig 2008-04-05 17:45:08.000000000 +0300 +++ sound/ppc/burgundy.c 2008-04-07 12:00:20.893307568 +0300 @@ -102,7 +102,8 @@ snd_pmac_burgundy_rcw(struct snd_pmac *c } static void -snd_pmac_burgundy_wcb(struct snd_pmac *chip, unsigned int addr, unsigned int val) +snd_pmac_burgundy_wcb(struct snd_pmac *chip, unsigned int addr, + unsigned int val) { out_le32(&chip->awacs->codec_ctrl, addr + 0x300000 + (val & 0xff)); snd_pmac_burgundy_busy_wait(chip); @@ -126,8 +127,11 @@ snd_pmac_burgundy_rcb(struct snd_pmac *c return val; } +#define BASE2ADDR(base) ((base) << 12) +#define ADDR2BASE(addr) ((addr) >> 12) + /* - * Burgundy volume: 0 - 100, stereo + * Burgundy volume: 0 - 100, stereo, word reg */ static void snd_pmac_burgundy_write_volume(struct snd_pmac *chip, unsigned int address, @@ -168,13 +172,6 @@ snd_pmac_burgundy_read_volume(struct snd volume[1] = 0; } - -/* - */ - -#define BASE2ADDR(base) ((base) << 12) -#define ADDR2BASE(addr) ((addr) >> 12) - static int snd_pmac_burgundy_info_volume(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -191,8 +188,8 @@ static int snd_pmac_burgundy_get_volume( struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); unsigned int addr = BASE2ADDR(kcontrol->private_value & 0xff); int shift = (kcontrol->private_value >> 8) & 0xff; - snd_pmac_burgundy_read_volume(chip, addr, ucontrol->value.integer.value, - shift); + snd_pmac_burgundy_read_volume(chip, addr, + ucontrol->value.integer.value, shift); return 0; } @@ -204,24 +201,163 @@ static int snd_pmac_burgundy_put_volume( int shift = (kcontrol->private_value >> 8) & 0xff; long nvoices[2]; - snd_pmac_burgundy_write_volume(chip, addr, ucontrol->value.integer.value, - shift); + snd_pmac_burgundy_write_volume(chip, addr, + ucontrol->value.integer.value, shift); snd_pmac_burgundy_read_volume(chip, addr, nvoices, shift); return (nvoices[0] != ucontrol->value.integer.value[0] || nvoices[1] != ucontrol->value.integer.value[1]); } -#define BURGUNDY_VOLUME(xname, xindex, addr, shift) \ +#define BURGUNDY_VOLUME_W(xname, xindex, addr, shift) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex,\ .info = snd_pmac_burgundy_info_volume,\ .get = snd_pmac_burgundy_get_volume,\ .put = snd_pmac_burgundy_put_volume,\ .private_value = ((ADDR2BASE(addr) & 0xff) | ((shift) << 8)) } -/* lineout/speaker */ +/* + * Burgundy volume: 0 - 100, stereo, 2-byte reg + */ +static void +snd_pmac_burgundy_write_volume_2b(struct snd_pmac *chip, unsigned int address, + long *volume, int off) +{ + int lvolume, rvolume; + + off |= off << 2; + lvolume = volume[0] ? volume[0] + BURGUNDY_VOLUME_OFFSET : 0; + rvolume = volume[1] ? volume[1] + BURGUNDY_VOLUME_OFFSET : 0; + + snd_pmac_burgundy_wcb(chip, address + off, lvolume); + snd_pmac_burgundy_wcb(chip, address + off + 0x500, rvolume); +} + +static void +snd_pmac_burgundy_read_volume_2b(struct snd_pmac *chip, unsigned int address, + long *volume, int off) +{ + volume[0] = snd_pmac_burgundy_rcb(chip, address + off); + if (volume[0] >= BURGUNDY_VOLUME_OFFSET) + volume[0] -= BURGUNDY_VOLUME_OFFSET; + else + volume[0] = 0; + volume[1] = snd_pmac_burgundy_rcb(chip, address + off + 0x100); + if (volume[1] >= BURGUNDY_VOLUME_OFFSET) + volume[1] -= BURGUNDY_VOLUME_OFFSET; + else + volume[1] = 0; +} + +static int snd_pmac_burgundy_info_volume_2b(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_pmac_burgundy_get_volume_2b(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); + unsigned int addr = BASE2ADDR(kcontrol->private_value & 0xff); + int off = kcontrol->private_value & 0x300; + snd_pmac_burgundy_read_volume_2b(chip, addr, + ucontrol->value.integer.value, off); + return 0; +} + +static int snd_pmac_burgundy_put_volume_2b(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); + unsigned int addr = BASE2ADDR(kcontrol->private_value & 0xff); + int off = kcontrol->private_value & 0x300; + long nvoices[2]; + + snd_pmac_burgundy_write_volume_2b(chip, addr, + ucontrol->value.integer.value, off); + snd_pmac_burgundy_read_volume_2b(chip, addr, nvoices, off); + return (nvoices[0] != ucontrol->value.integer.value[0] || + nvoices[1] != ucontrol->value.integer.value[1]); +} + +#define BURGUNDY_VOLUME_2B(xname, xindex, addr, off) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex,\ + .info = snd_pmac_burgundy_info_volume_2b,\ + .get = snd_pmac_burgundy_get_volume_2b,\ + .put = snd_pmac_burgundy_put_volume_2b,\ + .private_value = ((ADDR2BASE(addr) & 0xff) | ((off) << 8)) } + +/* + * Burgundy gain/attenuation: 0 - 15, mono/stereo, byte reg + */ +static int snd_pmac_burgundy_info_gain(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int stereo = (kcontrol->private_value >> 24) & 1; + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = stereo + 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 15; + return 0; +} + +static int snd_pmac_burgundy_get_gain(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); + unsigned int addr = BASE2ADDR(kcontrol->private_value & 0xff); + int stereo = (kcontrol->private_value >> 24) & 1; + int atten = (kcontrol->private_value >> 25) & 1; + int oval; + + oval = snd_pmac_burgundy_rcb(chip, addr); + if (atten) + oval = ~oval & 0xff; + ucontrol->value.integer.value[0] = oval & 0xf; + if (stereo) + ucontrol->value.integer.value[1] = (oval >> 4) & 0xf; + return 0; +} -static int snd_pmac_burgundy_info_switch_out(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) +static int snd_pmac_burgundy_put_gain(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); + unsigned int addr = BASE2ADDR(kcontrol->private_value & 0xff); + int stereo = (kcontrol->private_value >> 24) & 1; + int atten = (kcontrol->private_value >> 25) & 1; + int oval, val; + + oval = snd_pmac_burgundy_rcb(chip, addr); + if (atten) + oval = ~oval & 0xff; + val = ucontrol->value.integer.value[0]; + if (stereo) + val |= ucontrol->value.integer.value[1] << 4; + else + val |= ucontrol->value.integer.value[0] << 4; + if (atten) + val = ~val & 0xff; + snd_pmac_burgundy_wcb(chip, addr, val); + return val != oval; +} + +#define BURGUNDY_VOLUME_B(xname, xindex, addr, stereo, atten) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex,\ + .info = snd_pmac_burgundy_info_gain,\ + .get = snd_pmac_burgundy_get_gain,\ + .put = snd_pmac_burgundy_put_gain,\ + .private_value = (ADDR2BASE(addr) | ((stereo) << 24) | ((atten) << 25)) } + +/* + * Burgundy switch: 0/1, mono/stereo, word reg + */ +static int snd_pmac_burgundy_info_switch_w(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { int stereo = (kcontrol->private_value >> 24) & 1; uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; @@ -231,111 +367,207 @@ static int snd_pmac_burgundy_info_switch return 0; } -static int snd_pmac_burgundy_get_switch_out(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int snd_pmac_burgundy_get_switch_w(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); - int lmask = kcontrol->private_value & 0xff; - int rmask = (kcontrol->private_value >> 8) & 0xff; + unsigned int addr = BASE2ADDR((kcontrol->private_value >> 16) & 0xff); + int lmask = 1 << (kcontrol->private_value & 0xff); + int rmask = 1 << ((kcontrol->private_value >> 8) & 0xff); int stereo = (kcontrol->private_value >> 24) & 1; - int val = snd_pmac_burgundy_rcb(chip, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES); + int val = snd_pmac_burgundy_rcw(chip, addr); ucontrol->value.integer.value[0] = (val & lmask) ? 1 : 0; if (stereo) ucontrol->value.integer.value[1] = (val & rmask) ? 1 : 0; return 0; } -static int snd_pmac_burgundy_put_switch_out(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int snd_pmac_burgundy_put_switch_w(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); - int lmask = kcontrol->private_value & 0xff; - int rmask = (kcontrol->private_value >> 8) & 0xff; + unsigned int addr = BASE2ADDR((kcontrol->private_value >> 16) & 0xff); + int lmask = 1 << (kcontrol->private_value & 0xff); + int rmask = 1 << ((kcontrol->private_value >> 8) & 0xff); int stereo = (kcontrol->private_value >> 24) & 1; int val, oval; - oval = snd_pmac_burgundy_rcb(chip, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES); - val = oval & ~(lmask | rmask); + oval = snd_pmac_burgundy_rcw(chip, addr); + val = oval & ~(lmask | (stereo ? rmask : 0)); if (ucontrol->value.integer.value[0]) val |= lmask; if (stereo && ucontrol->value.integer.value[1]) val |= rmask; - snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, val); + snd_pmac_burgundy_wcw(chip, addr, val); return val != oval; } -#define BURGUNDY_OUTPUT_SWITCH(xname, xindex, lmask, rmask, stereo) \ +#define BURGUNDY_SWITCH_W(xname, xindex, addr, lbit, rbit, stereo) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex,\ - .info = snd_pmac_burgundy_info_switch_out,\ - .get = snd_pmac_burgundy_get_switch_out,\ - .put = snd_pmac_burgundy_put_switch_out,\ - .private_value = ((lmask) | ((rmask) << 8) | ((stereo) << 24)) } - -/* line/speaker output volume */ -static int snd_pmac_burgundy_info_volume_out(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) + .info = snd_pmac_burgundy_info_switch_w,\ + .get = snd_pmac_burgundy_get_switch_w,\ + .put = snd_pmac_burgundy_put_switch_w,\ + .private_value = ((lbit) | ((rbit) << 8)\ + | (ADDR2BASE(addr) << 16) | ((stereo) << 24)) } + +/* + * Burgundy switch: 0/1, mono/stereo, byte reg, bit mask + */ +static int snd_pmac_burgundy_info_switch_b(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { int stereo = (kcontrol->private_value >> 24) & 1; - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; uinfo->count = stereo + 1; uinfo->value.integer.min = 0; - uinfo->value.integer.max = 15; + uinfo->value.integer.max = 1; return 0; } -static int snd_pmac_burgundy_get_volume_out(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int snd_pmac_burgundy_get_switch_b(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); - unsigned int addr = BASE2ADDR(kcontrol->private_value & 0xff); + unsigned int addr = BASE2ADDR((kcontrol->private_value >> 16) & 0xff); + int lmask = kcontrol->private_value & 0xff; + int rmask = (kcontrol->private_value >> 8) & 0xff; int stereo = (kcontrol->private_value >> 24) & 1; - int oval; - - oval = ~snd_pmac_burgundy_rcb(chip, addr) & 0xff; - ucontrol->value.integer.value[0] = oval & 0xf; + int val = snd_pmac_burgundy_rcb(chip, addr); + ucontrol->value.integer.value[0] = (val & lmask) ? 1 : 0; if (stereo) - ucontrol->value.integer.value[1] = (oval >> 4) & 0xf; + ucontrol->value.integer.value[1] = (val & rmask) ? 1 : 0; return 0; } -static int snd_pmac_burgundy_put_volume_out(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int snd_pmac_burgundy_put_switch_b(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); - unsigned int addr = BASE2ADDR(kcontrol->private_value & 0xff); + unsigned int addr = BASE2ADDR((kcontrol->private_value >> 16) & 0xff); + int lmask = kcontrol->private_value & 0xff; + int rmask = (kcontrol->private_value >> 8) & 0xff; int stereo = (kcontrol->private_value >> 24) & 1; - unsigned int oval, val; - - oval = ~snd_pmac_burgundy_rcb(chip, addr) & 0xff; - val = ucontrol->value.integer.value[0] & 15; - if (stereo) - val |= (ucontrol->value.integer.value[1] & 15) << 4; - else - val |= val << 4; - val = ~val & 0xff; + int val, oval; + oval = snd_pmac_burgundy_rcb(chip, addr); + val = oval & ~(lmask | rmask); + if (ucontrol->value.integer.value[0]) + val |= lmask; + if (stereo && ucontrol->value.integer.value[1]) + val |= rmask; snd_pmac_burgundy_wcb(chip, addr, val); return val != oval; } -#define BURGUNDY_OUTPUT_VOLUME(xname, xindex, addr, stereo) \ +#define BURGUNDY_SWITCH_B(xname, xindex, addr, lmask, rmask, stereo) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex,\ - .info = snd_pmac_burgundy_info_volume_out,\ - .get = snd_pmac_burgundy_get_volume_out,\ - .put = snd_pmac_burgundy_put_volume_out,\ - .private_value = (ADDR2BASE(addr) | ((stereo) << 24)) } + .info = snd_pmac_burgundy_info_switch_b,\ + .get = snd_pmac_burgundy_get_switch_b,\ + .put = snd_pmac_burgundy_put_switch_b,\ + .private_value = ((lmask) | ((rmask) << 8)\ + | (ADDR2BASE(addr) << 16) | ((stereo) << 24)) } +/* + * Burgundy mixers + */ static struct snd_kcontrol_new snd_pmac_burgundy_mixers[] __initdata = { - BURGUNDY_VOLUME("Master Playback Volume", 0, MASK_ADDR_BURGUNDY_MASTER_VOLUME, 8), - BURGUNDY_VOLUME("Line Playback Volume", 0, MASK_ADDR_BURGUNDY_VOLLINE, 16), - BURGUNDY_VOLUME("CD Playback Volume", 0, MASK_ADDR_BURGUNDY_VOLCD, 16), - BURGUNDY_VOLUME("Mic Playback Volume", 0, MASK_ADDR_BURGUNDY_VOLMIC, 16), - BURGUNDY_OUTPUT_VOLUME("PC Speaker Playback Volume", 0, MASK_ADDR_BURGUNDY_ATTENHP, 0), - /*BURGUNDY_OUTPUT_VOLUME("PCM Playback Volume", 0, MASK_ADDR_BURGUNDY_ATTENLINEOUT, 1),*/ - BURGUNDY_OUTPUT_VOLUME("Headphone Playback Volume", 0, MASK_ADDR_BURGUNDY_ATTENSPEAKER, 1), -}; -static struct snd_kcontrol_new snd_pmac_burgundy_master_sw __initdata = -BURGUNDY_OUTPUT_SWITCH("Headphone Playback Switch", 0, BURGUNDY_OUTPUT_LEFT, BURGUNDY_OUTPUT_RIGHT, 1); -static struct snd_kcontrol_new snd_pmac_burgundy_speaker_sw __initdata = -BURGUNDY_OUTPUT_SWITCH("PC Speaker Playback Switch", 0, BURGUNDY_OUTPUT_INTERN, 0, 0); + BURGUNDY_VOLUME_W("Master Playback Volume", 0, + MASK_ADDR_BURGUNDY_MASTER_VOLUME, 8), + BURGUNDY_VOLUME_W("CD Capture Volume", 0, + MASK_ADDR_BURGUNDY_VOLCD, 16), + BURGUNDY_VOLUME_2B("Input Capture Volume", 0, + MASK_ADDR_BURGUNDY_VOLMIX01, 2), + BURGUNDY_VOLUME_2B("Mixer Playback Volume", 0, + MASK_ADDR_BURGUNDY_VOLMIX23, 0), + BURGUNDY_VOLUME_B("CD Gain Capture Volume", 0, + MASK_ADDR_BURGUNDY_GAINCD, 1, 0), + BURGUNDY_SWITCH_W("Master Capture Switch", 0, + MASK_ADDR_BURGUNDY_OUTPUTENABLES, 24, 0, 0), + BURGUNDY_SWITCH_W("CD Capture Switch", 0, + MASK_ADDR_BURGUNDY_CAPTURESELECTS, 0, 16, 1), + BURGUNDY_SWITCH_W("CD Playback Switch", 0, + MASK_ADDR_BURGUNDY_OUTPUTSELECTS, 0, 16, 1), +/* BURGUNDY_SWITCH_W("Loop Capture Switch", 0, + * MASK_ADDR_BURGUNDY_CAPTURESELECTS, 8, 24, 1), + * BURGUNDY_SWITCH_B("Mixer out Capture Switch", 0, + * MASK_ADDR_BURGUNDY_HOSTIFAD, 0x02, 0, 0), + * BURGUNDY_SWITCH_B("Mixer Capture Switch", 0, + * MASK_ADDR_BURGUNDY_HOSTIFAD, 0x01, 0, 0), + * BURGUNDY_SWITCH_B("PCM out Capture Switch", 0, + * MASK_ADDR_BURGUNDY_HOSTIFEH, 0x02, 0, 0), + */ BURGUNDY_SWITCH_B("PCM Capture Switch", 0, + MASK_ADDR_BURGUNDY_HOSTIFEH, 0x01, 0, 0) +}; +static struct snd_kcontrol_new snd_pmac_burgundy_mixers_imac[] __initdata = { + BURGUNDY_VOLUME_W("Line in Capture Volume", 0, + MASK_ADDR_BURGUNDY_VOLLINE, 16), + BURGUNDY_VOLUME_W("Mic Capture Volume", 0, + MASK_ADDR_BURGUNDY_VOLMIC, 16), + BURGUNDY_VOLUME_B("Line in Gain Capture Volume", 0, + MASK_ADDR_BURGUNDY_GAINLINE, 1, 0), + BURGUNDY_VOLUME_B("Mic Gain Capture Volume", 0, + MASK_ADDR_BURGUNDY_GAINMIC, 1, 0), + BURGUNDY_VOLUME_B("PC Speaker Playback Volume", 0, + MASK_ADDR_BURGUNDY_ATTENSPEAKER, 1, 1), + BURGUNDY_VOLUME_B("Line out Playback Volume", 0, + MASK_ADDR_BURGUNDY_ATTENLINEOUT, 1, 1), + BURGUNDY_VOLUME_B("Headphone Playback Volume", 0, + MASK_ADDR_BURGUNDY_ATTENHP, 1, 1), + BURGUNDY_SWITCH_W("Line in Capture Switch", 0, + MASK_ADDR_BURGUNDY_CAPTURESELECTS, 1, 17, 1), + BURGUNDY_SWITCH_W("Mic Capture Switch", 0, + MASK_ADDR_BURGUNDY_CAPTURESELECTS, 2, 18, 1), + BURGUNDY_SWITCH_W("Line in Playback Switch", 0, + MASK_ADDR_BURGUNDY_OUTPUTSELECTS, 1, 17, 1), + BURGUNDY_SWITCH_W("Mic Playback Switch", 0, + MASK_ADDR_BURGUNDY_OUTPUTSELECTS, 2, 18, 1), + BURGUNDY_SWITCH_B("Mic Boost Capture Switch", 0, + MASK_ADDR_BURGUNDY_INPBOOST, 0x40, 0x80, 1) +}; +static struct snd_kcontrol_new snd_pmac_burgundy_mixers_pmac[] __initdata = { + BURGUNDY_VOLUME_W("Line in Capture Volume", 0, + MASK_ADDR_BURGUNDY_VOLMIC, 16), + BURGUNDY_VOLUME_B("Line in Gain Capture Volume", 0, + MASK_ADDR_BURGUNDY_GAINMIC, 1, 0), + BURGUNDY_VOLUME_B("PC Speaker Playback Volume", 0, + MASK_ADDR_BURGUNDY_ATTENMONO, 0, 1), + BURGUNDY_VOLUME_B("Line out Playback Volume", 0, + MASK_ADDR_BURGUNDY_ATTENSPEAKER, 1, 1), + BURGUNDY_SWITCH_W("Line in Capture Switch", 0, + MASK_ADDR_BURGUNDY_CAPTURESELECTS, 2, 18, 1), + BURGUNDY_SWITCH_W("Line in Playback Switch", 0, + MASK_ADDR_BURGUNDY_OUTPUTSELECTS, 2, 18, 1), +/* BURGUNDY_SWITCH_B("Line in Boost Capture Switch", 0, + * MASK_ADDR_BURGUNDY_INPBOOST, 0x40, 0x80, 1) */ +}; +static struct snd_kcontrol_new snd_pmac_burgundy_master_sw_imac __initdata = +BURGUNDY_SWITCH_B("Master Playback Switch", 0, + MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, + BURGUNDY_OUTPUT_LEFT | BURGUNDY_LINEOUT_LEFT | BURGUNDY_HP_LEFT, + BURGUNDY_OUTPUT_RIGHT | BURGUNDY_LINEOUT_RIGHT | BURGUNDY_HP_RIGHT, 1); +static struct snd_kcontrol_new snd_pmac_burgundy_master_sw_pmac __initdata = +BURGUNDY_SWITCH_B("Master Playback Switch", 0, + MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, + BURGUNDY_OUTPUT_INTERN + | BURGUNDY_OUTPUT_LEFT, BURGUNDY_OUTPUT_RIGHT, 1); +static struct snd_kcontrol_new snd_pmac_burgundy_speaker_sw_imac __initdata = +BURGUNDY_SWITCH_B("PC Speaker Playback Switch", 0, + MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, + BURGUNDY_OUTPUT_LEFT, BURGUNDY_OUTPUT_RIGHT, 1); +static struct snd_kcontrol_new snd_pmac_burgundy_speaker_sw_pmac __initdata = +BURGUNDY_SWITCH_B("PC Speaker Playback Switch", 0, + MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, + BURGUNDY_OUTPUT_INTERN, 0, 0); +static struct snd_kcontrol_new snd_pmac_burgundy_line_sw_imac __initdata = +BURGUNDY_SWITCH_B("Line out Playback Switch", 0, + MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, + BURGUNDY_LINEOUT_LEFT, BURGUNDY_LINEOUT_RIGHT, 1); +static struct snd_kcontrol_new snd_pmac_burgundy_line_sw_pmac __initdata = +BURGUNDY_SWITCH_B("Line out Playback Switch", 0, + MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, + BURGUNDY_OUTPUT_LEFT, BURGUNDY_OUTPUT_RIGHT, 1); +static struct snd_kcontrol_new snd_pmac_burgundy_hp_sw_imac __initdata = +BURGUNDY_SWITCH_B("Headphone Playback Switch", 0, + MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, + BURGUNDY_HP_LEFT, BURGUNDY_HP_RIGHT, 1); #ifdef PMAC_SUPPORT_AUTOMUTE @@ -347,19 +579,30 @@ static int snd_pmac_burgundy_detect_head return (in_le32(&chip->awacs->codec_stat) & chip->hp_stat_mask) ? 1 : 0; } -static void snd_pmac_burgundy_update_automute(struct snd_pmac *chip, int do_notify) +static void snd_pmac_burgundy_update_automute(struct snd_pmac *chip, + int do_notify) { if (chip->auto_mute) { + int imac = machine_is_compatible("iMac"); int reg, oreg; - reg = oreg = snd_pmac_burgundy_rcb(chip, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES); - reg &= ~(BURGUNDY_OUTPUT_LEFT | BURGUNDY_OUTPUT_RIGHT | BURGUNDY_OUTPUT_INTERN); + reg = oreg = snd_pmac_burgundy_rcb(chip, + MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES); + reg &= imac ? ~(BURGUNDY_OUTPUT_LEFT | BURGUNDY_OUTPUT_RIGHT + | BURGUNDY_HP_LEFT | BURGUNDY_HP_RIGHT) + : ~(BURGUNDY_OUTPUT_LEFT | BURGUNDY_OUTPUT_RIGHT + | BURGUNDY_OUTPUT_INTERN); if (snd_pmac_burgundy_detect_headphone(chip)) - reg |= BURGUNDY_OUTPUT_LEFT | BURGUNDY_OUTPUT_RIGHT; + reg |= imac ? (BURGUNDY_HP_LEFT | BURGUNDY_HP_RIGHT) + : (BURGUNDY_OUTPUT_LEFT + | BURGUNDY_OUTPUT_RIGHT); else - reg |= BURGUNDY_OUTPUT_INTERN; + reg |= imac ? (BURGUNDY_OUTPUT_LEFT + | BURGUNDY_OUTPUT_RIGHT) + : (BURGUNDY_OUTPUT_INTERN); if (do_notify && reg == oreg) return; - snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, reg); + snd_pmac_burgundy_wcb(chip, + MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, reg); if (do_notify) { snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->master_sw_ctl->id); @@ -378,6 +621,7 @@ static void snd_pmac_burgundy_update_aut */ int __init snd_pmac_burgundy_init(struct snd_pmac *chip) { + int imac = machine_is_compatible("iMac"); int i, err; /* Checks to see the chip is alive and kicking */ @@ -386,7 +630,7 @@ int __init snd_pmac_burgundy_init(struct return 1; } - snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_OUTPUTENABLES, + snd_pmac_burgundy_wcw(chip, MASK_ADDR_BURGUNDY_OUTPUTENABLES, DEF_BURGUNDY_OUTPUTENABLES); snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, DEF_BURGUNDY_MORE_OUTPUTENABLES); @@ -396,7 +640,8 @@ int __init snd_pmac_burgundy_init(struct snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_INPSEL21, DEF_BURGUNDY_INPSEL21); snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_INPSEL3, - DEF_BURGUNDY_INPSEL3); + imac ? DEF_BURGUNDY_INPSEL3_IMAC + : DEF_BURGUNDY_INPSEL3_PMAC); snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_GAINCD, DEF_BURGUNDY_GAINCD); snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_GAINLINE, @@ -422,23 +667,51 @@ int __init snd_pmac_burgundy_init(struct snd_pmac_burgundy_wcw(chip, MASK_ADDR_BURGUNDY_VOLMIC, DEF_BURGUNDY_VOLMIC); - if (chip->hp_stat_mask == 0) + if (chip->hp_stat_mask == 0) { /* set headphone-jack detection bit */ - chip->hp_stat_mask = 0x04; - + if (imac) + chip->hp_stat_mask = BURGUNDY_HPDETECT_IMAC_UPPER + | BURGUNDY_HPDETECT_IMAC_LOWER + | BURGUNDY_HPDETECT_IMAC_SIDE; + else + chip->hp_stat_mask = BURGUNDY_HPDETECT_PMAC_BACK; + } /* * build burgundy mixers */ strcpy(chip->card->mixername, "PowerMac Burgundy"); for (i = 0; i < ARRAY_SIZE(snd_pmac_burgundy_mixers); i++) { - if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_pmac_burgundy_mixers[i], chip))) < 0) + if ((err = snd_ctl_add(chip->card, + snd_ctl_new1(&snd_pmac_burgundy_mixers[i], chip))) < 0) + return err; + } + for (i = 0; i < (imac ? ARRAY_SIZE(snd_pmac_burgundy_mixers_imac) + : ARRAY_SIZE(snd_pmac_burgundy_mixers_pmac)); i++) { + if ((err = snd_ctl_add(chip->card, + snd_ctl_new1(imac ? &snd_pmac_burgundy_mixers_imac[i] + : &snd_pmac_burgundy_mixers_pmac[i], chip))) < 0) return err; } - chip->master_sw_ctl = snd_ctl_new1(&snd_pmac_burgundy_master_sw, chip); + chip->master_sw_ctl = snd_ctl_new1(imac + ? &snd_pmac_burgundy_master_sw_imac + : &snd_pmac_burgundy_master_sw_pmac, chip); if ((err = snd_ctl_add(chip->card, chip->master_sw_ctl)) < 0) return err; - chip->speaker_sw_ctl = snd_ctl_new1(&snd_pmac_burgundy_speaker_sw, chip); + chip->master_sw_ctl = snd_ctl_new1(imac + ? &snd_pmac_burgundy_line_sw_imac + : &snd_pmac_burgundy_line_sw_pmac, chip); + if ((err = snd_ctl_add(chip->card, chip->master_sw_ctl)) < 0) + return err; + if (imac) { + chip->master_sw_ctl = snd_ctl_new1( + &snd_pmac_burgundy_hp_sw_imac, chip); + if ((err = snd_ctl_add(chip->card, chip->master_sw_ctl)) < 0) + return err; + } + chip->speaker_sw_ctl = snd_ctl_new1(imac + ? &snd_pmac_burgundy_speaker_sw_imac + : &snd_pmac_burgundy_speaker_sw_pmac, chip); if ((err = snd_ctl_add(chip->card, chip->speaker_sw_ctl)) < 0) return err; #ifdef PMAC_SUPPORT_AUTOMUTE --- sound/ppc/burgundy.h.orig 2008-04-05 17:45:08.000000000 +0300 +++ sound/ppc/burgundy.h 2008-04-05 22:02:44.000000000 +0300 @@ -22,6 +22,7 @@ #ifndef __BURGUNDY_H #define __BURGUNDY_H +#define MASK_ADDR_BURGUNDY_INPBOOST (0x10 << 12) #define MASK_ADDR_BURGUNDY_INPSEL21 (0x11 << 12) #define MASK_ADDR_BURGUNDY_INPSEL3 (0x12 << 12) @@ -35,7 +36,10 @@ #define MASK_ADDR_BURGUNDY_VOLCH3 (0x22 << 12) #define MASK_ADDR_BURGUNDY_VOLCH4 (0x23 << 12) +#define MASK_ADDR_BURGUNDY_CAPTURESELECTS (0x2A << 12) #define MASK_ADDR_BURGUNDY_OUTPUTSELECTS (0x2B << 12) +#define MASK_ADDR_BURGUNDY_VOLMIX01 (0x2D << 12) +#define MASK_ADDR_BURGUNDY_VOLMIX23 (0x2E << 12) #define MASK_ADDR_BURGUNDY_OUTPUTENABLES (0x2F << 12) #define MASK_ADDR_BURGUNDY_MASTER_VOLUME (0x30 << 12) @@ -45,6 +49,10 @@ #define MASK_ADDR_BURGUNDY_ATTENSPEAKER (0x62 << 12) #define MASK_ADDR_BURGUNDY_ATTENLINEOUT (0x63 << 12) #define MASK_ADDR_BURGUNDY_ATTENHP (0x64 << 12) +#define MASK_ADDR_BURGUNDY_ATTENMONO (0x65 << 12) + +#define MASK_ADDR_BURGUNDY_HOSTIFAD (0x78 << 12) +#define MASK_ADDR_BURGUNDY_HOSTIFEH (0x79 << 12) #define MASK_ADDR_BURGUNDY_VOLCD (MASK_ADDR_BURGUNDY_VOLCH1) #define MASK_ADDR_BURGUNDY_VOLLINE (MASK_ADDR_BURGUNDY_VOLCH2) @@ -59,21 +67,22 @@ /* These are all default values for the burgundy */ #define DEF_BURGUNDY_INPSEL21 (0xAA) -#define DEF_BURGUNDY_INPSEL3 (0x0A) +#define DEF_BURGUNDY_INPSEL3_IMAC (0x0A) +#define DEF_BURGUNDY_INPSEL3_PMAC (0x05) #define DEF_BURGUNDY_GAINCD (0x33) #define DEF_BURGUNDY_GAINLINE (0x44) #define DEF_BURGUNDY_GAINMIC (0x44) #define DEF_BURGUNDY_GAINMODEM (0x06) -/* Remember: lowest volume here is 0x9b */ +/* Remember: lowest volume here is 0x9B (155) */ #define DEF_BURGUNDY_VOLCD (0xCCCCCCCC) #define DEF_BURGUNDY_VOLLINE (0x00000000) #define DEF_BURGUNDY_VOLMIC (0x00000000) #define DEF_BURGUNDY_VOLMODEM (0xCCCCCCCC) -#define DEF_BURGUNDY_OUTPUTSELECTS (0x010f010f) -#define DEF_BURGUNDY_OUTPUTENABLES (0x0A) +#define DEF_BURGUNDY_OUTPUTSELECTS (0x010F010F) +#define DEF_BURGUNDY_OUTPUTENABLES (0x0100000A) /* #define DEF_BURGUNDY_MASTER_VOLUME (0xFFFFFFFF) */ /* too loud */ #define DEF_BURGUNDY_MASTER_VOLUME (0xDDDDDDDD) @@ -84,12 +93,22 @@ #define DEF_BURGUNDY_ATTENLINEOUT (0xCC) #define DEF_BURGUNDY_ATTENHP (0xCC) -/* OUTPUTENABLES bits */ +/* MORE_OUTPUTENABLES bits */ #define BURGUNDY_OUTPUT_LEFT 0x02 #define BURGUNDY_OUTPUT_RIGHT 0x04 +#define BURGUNDY_LINEOUT_LEFT 0x08 +#define BURGUNDY_LINEOUT_RIGHT 0x10 +#define BURGUNDY_HP_LEFT 0x20 +#define BURGUNDY_HP_RIGHT 0x40 #define BURGUNDY_OUTPUT_INTERN 0x80 -/* volume offset */ +/* Headphone detection bits */ +#define BURGUNDY_HPDETECT_PMAC_BACK 0x04 +#define BURGUNDY_HPDETECT_IMAC_SIDE 0x04 +#define BURGUNDY_HPDETECT_IMAC_UPPER 0x08 +#define BURGUNDY_HPDETECT_IMAC_LOWER 0x01 + +/* Volume offset */ #define BURGUNDY_VOLUME_OFFSET 155 #endif /* __BURGUNDY_H */

1 0

[alsa-devel] Missing patches for 2.6.26 kernel?
by Takashi Iwai 14 Apr '08

14 Apr '08

Hi all, I'm just back from vacation and turned to be a patch monkey now. Already applied many pending patches as much as I catched up. If you have patches that are still missing in the upstream tree, please let me know promptly so that they are timely on 2.6.26 merge window. Thanks in advance, Takashi

1 0

[alsa-devel] [PATCH] snd_usb_caiaq: make high sample rates work with A8DJ
by Daniel Mack 14 Apr '08

14 Apr '08

This patch for snd_usb_caiaq makes sample rates higher dann 48KHz work with devices which have more than 2 stereo input/output pairs. Signed-off-by: Daniel Mack <daniel(a)caiaq.de>

2 1

[alsa-devel] [PATCH] ASoC: Add support for 19.2 MHz MCLK in TLV320AIC3X
by Mark Brown 14 Apr '08

14 Apr '08

From: Jarkko Nikula <jarkko.nikula(a)nokia.com> Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/tlv320aic3x.c | 11 +++++++++++ 1 files changed, 11 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index e6a730b..630684f 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -661,42 +661,52 @@ struct aic3x_rate_divs { static const struct aic3x_rate_divs aic3x_divs[] = { /* 8k */ {12000000, 8000, 48000, 0xa, 16, 3840}, + {19200000, 8000, 48000, 0xa, 10, 2400}, {22579200, 8000, 48000, 0xa, 8, 7075}, {33868800, 8000, 48000, 0xa, 5, 8049}, /* 11.025k */ {12000000, 11025, 44100, 0x6, 15, 528}, + {19200000, 11025, 44100, 0x6, 9, 4080}, {22579200, 11025, 44100, 0x6, 8, 0}, {33868800, 11025, 44100, 0x6, 5, 3333}, /* 16k */ {12000000, 16000, 48000, 0x4, 16, 3840}, + {19200000, 16000, 48000, 0x4, 10, 2400}, {22579200, 16000, 48000, 0x4, 8, 7075}, {33868800, 16000, 48000, 0x4, 5, 8049}, /* 22.05k */ {12000000, 22050, 44100, 0x2, 15, 528}, + {19200000, 22050, 44100, 0x2, 9, 4080}, {22579200, 22050, 44100, 0x2, 8, 0}, {33868800, 22050, 44100, 0x2, 5, 3333}, /* 32k */ {12000000, 32000, 48000, 0x1, 16, 3840}, + {19200000, 32000, 48000, 0x1, 10, 2400}, {22579200, 32000, 48000, 0x1, 8, 7075}, {33868800, 32000, 48000, 0x1, 5, 8049}, /* 44.1k */ {12000000, 44100, 44100, 0x0, 15, 528}, + {19200000, 44100, 44100, 0x0, 9, 4080}, {22579200, 44100, 44100, 0x0, 8, 0}, {33868800, 44100, 44100, 0x0, 5, 3333}, /* 48k */ {12000000, 48000, 48000, 0x0, 16, 3840}, + {19200000, 48000, 48000, 0x0, 10, 2400}, {22579200, 48000, 48000, 0x0, 8, 7075}, {33868800, 48000, 48000, 0x0, 5, 8049}, /* 64k */ {12000000, 64000, 96000, 0x1, 16, 3840}, + {19200000, 64000, 96000, 0x1, 10, 2400}, {22579200, 64000, 96000, 0x1, 8, 7075}, {33868800, 64000, 96000, 0x1, 5, 8049}, /* 88.2k */ {12000000, 88200, 88200, 0x0, 15, 528}, + {19200000, 88200, 88200, 0x0, 9, 4080}, {22579200, 88200, 88200, 0x0, 8, 0}, {33868800, 88200, 88200, 0x0, 5, 3333}, /* 96k */ {12000000, 96000, 96000, 0x0, 16, 3840}, + {19200000, 96000, 96000, 0x0, 10, 2400}, {22579200, 96000, 96000, 0x0, 8, 7075}, {33868800, 96000, 96000, 0x0, 5, 8049}, }; @@ -818,6 +828,7 @@ static int aic3x_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, switch (freq) { case 12000000: + case 19200000: case 22579200: case 33868800: aic3x->sysclk = freq; -- 1.5.5

2 1

[alsa-devel] [PATCH 1/2] soc - Fix s3c24xx-i2s LR sync while timer ticks are disabled
by Mark Brown 14 Apr '08

14 Apr '08

From: Werner Almesberger <werner(a)openmoko.org> When timer ticks are disabled when calling sound/soc/s3c24xx/s3c24xx-i2s.c:s3c24xx_snd_lrsync and the LR signal never happens, we loop forever. This has been observed in the following call chain: snd_pcm_common_ioctl1 -> snd_pcm_action_lock_irq -> snd_pcm_action_single -> snd_pcm_do_resume -> soc_pcm_trigger -> s3c24xx_i2s_trigger The patch below changes the timeout mechanism to use udelay, which doesn't need timer ticks. Signed-off-by: Werner Almesberger <werner(a)openmoko.org> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/s3c24xx/s3c24xx-i2s.c | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 0a3c630..6b3cbbf 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -180,7 +180,7 @@ static void s3c24xx_snd_rxctrl(int on) static int s3c24xx_snd_lrsync(void) { u32 iiscon; - unsigned long timeout = jiffies + msecs_to_jiffies(5); + int timeout = 50; /* 5ms */ DBG("Entered %s\n", __FUNCTION__); @@ -189,8 +189,9 @@ static int s3c24xx_snd_lrsync(void) if (iiscon & S3C2410_IISCON_LRINDEX) break; - if (time_after(jiffies, timeout)) + if (!timeout--) return -ETIMEDOUT; + udelay(100); } return 0; -- 1.5.5

2 3

[alsa-devel] [PATCH] hda-intel: Add Quanta IL1 ALC267 model
by Herton Ronaldo Krzesinski 14 Apr '08

14 Apr '08

This adds support for Quanta IL1 mini-notebook to alsa, defining a new model for it. It comes with an ALC267 codec chip. Some notes about this model: * In headphone automute, I use AC_VERB_SET_PIN_WIDGET_CONTROL instead of common amp mute, to avoid conflict with mixer switch (mixer and automute use the same nid). * The only connected capture sources in the hardware are the internal mic and external mic jack. So instead of using an input source selector like on other ALC268 models, the mic automute automatically switch between captures. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com> diff -p -up alsa-kernel/Documentation/ALSA-Configuration.txt.orig alsa-kernel/Documentation/ALSA-Configuration.txt --- alsa-kernel/Documentation/ALSA-Configuration.txt.orig 2008-03-25 16:30:07.000000000 -0300 +++ alsa-kernel/Documentation/ALSA-Configuration.txt 2008-03-25 16:33:52.000000000 -0300 @@ -829,7 +829,8 @@ Prior to version 0.9.0rc4 options had a basic fixed pin assignment w/o SPDIF auto auto-config reading BIOS (default) - ALC268 + ALC267/268 + quanta-il1 Quanta IL1 mini-notebook 3stack 3-stack model toshiba Toshiba A205 acer Acer laptops diff -p -up alsa-kernel/pci/hda/patch_realtek.c.orig alsa-kernel/pci/hda/patch_realtek.c --- alsa-kernel/pci/hda/patch_realtek.c.orig 2008-03-25 16:30:07.000000000 -0300 +++ alsa-kernel/pci/hda/patch_realtek.c 2008-03-25 16:33:52.000000000 -0300 @@ -105,6 +105,7 @@ enum { /* ALC268 models */ enum { + ALC267_QUANTA_IL1, ALC268_3ST, ALC268_TOSHIBA, ALC268_ACER, @@ -9896,6 +9897,64 @@ static void alc268_dell_unsol_event(stru #define alc268_dell_init_hook alc268_dell_automute +static struct snd_kcontrol_new alc267_quanta_il1_mixer[] = { + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x2, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Capture Volume", 0x23, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Mic Capture Switch", 0x23, 2, HDA_OUTPUT), + HDA_CODEC_VOLUME("Ext Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), + { } +}; + +static struct hda_verb alc267_quanta_il1_verbs[] = { + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN}, + { } +}; + +static void alc267_quanta_il1_hp_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + present ? 0 : PIN_OUT); +} + +static void alc267_quanta_il1_mic_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x18, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_write(codec, 0x23, 0, + AC_VERB_SET_CONNECT_SEL, + present ? 0x00 : 0x01); +} + +static void alc267_quanta_il1_automute(struct hda_codec *codec) +{ + alc267_quanta_il1_hp_automute(codec); + alc267_quanta_il1_mic_automute(codec); +} + +static void alc267_quanta_il1_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc267_quanta_il1_hp_automute(codec); + break; + case ALC880_MIC_EVENT: + alc267_quanta_il1_mic_automute(codec); + break; + } +} + /* * generic initialization of ADC, input mixers and output mixers */ @@ -10318,6 +10377,7 @@ static void alc268_auto_init(struct hda_ * configuration and preset */ static const char *alc268_models[ALC268_MODEL_LAST] = { + [ALC267_QUANTA_IL1] = "quanta-il1", [ALC268_3ST] = "3stack", [ALC268_TOSHIBA] = "toshiba", [ALC268_ACER] = "acer", @@ -10340,11 +10400,27 @@ static struct snd_pci_quirk alc268_cfg_t SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA), SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA), SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER), + SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1), SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO), {} }; static struct alc_config_preset alc268_presets[] = { + [ALC267_QUANTA_IL1] = { + .mixers = { alc267_quanta_il1_mixer }, + .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, + alc267_quanta_il1_verbs }, + .num_dacs = ARRAY_SIZE(alc268_dac_nids), + .dac_nids = alc268_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), + .adc_nids = alc268_adc_nids_alt, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc268_modes), + .channel_mode = alc268_modes, + .input_mux = &alc268_capture_source, + .unsol_event = alc267_quanta_il1_unsol_event, + .init_hook = alc267_quanta_il1_automute, + }, [ALC268_3ST] = { .mixers = { alc268_base_mixer, alc268_capture_alt_mixer, alc268_beep_mixer }, -- []'s Herton

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Re: [alsa-devel] [PATCH] es1968: fix jitter on some maestro cards
by Rene Herman 14 Apr '08

14 Apr '08

On 30-03-08 00:49, Andreas Mueller wrote: > This patch suppresses jitter on several Maestro cards in stereo mode > (ALSA of course). > > The patch is also incorporated in the *BSD drivers where I "ported" it from. > I am barely used to driver-development so please take a look (esp. the channel > parameters), nevertheless it completely works for me. > > Without this patch most of the stereo audio gets out of sync and really > distorted (oss-emulation with mplayer at 48000khz worked somehow). > > CCed to those listed in the .c file. Added the alsa-devel list (and re-attached the patch). My TerraTec DMX (ESS Canyon3D, ES1970MS-3D) doesn't need this but doesn't seem to suffer from it either. Otherwise no idea about this. Rene. --- sound/pci/es1968.c.old 2008-03-29 19:17:16.771116317 +0100 +++ sound/pci/es1968.c 2008-03-30 00:14:35.363057160 +0100 @@ -1816,7 +1816,24 @@ return 0; } +/* + * suppress jitter on some maestros when playing stereo + */ +static void snd_es1968_suppress_jitter(struct es1968 *chip, struct esschan *es) +{ + unsigned int cp1; + unsigned int cp2; + unsigned int diff; + cp1 = __apu_get_register(chip, 0, 5); + cp2 = __apu_get_register(chip, 1, 5); + diff = (cp1 > cp2 ? cp1 - cp2 : cp2 - cp1); + + if (diff > 1) { + __maestro_write(chip, IDR0_DATA_PORT, cp1); + } +} + /* * update pointer */ @@ -1937,8 +1954,11 @@ struct esschan *es; spin_lock(&chip->substream_lock); list_for_each_entry(es, &chip->substream_list, list) { - if (es->running) + if (es->running) { snd_es1968_update_pcm(chip, es); + if (es->fmt & ESS_FMT_STEREO) + snd_es1968_suppress_jitter(chip, es); + } } spin_unlock(&chip->substream_lock); if (chip->in_measurement) {

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[alsa-devel] [PATCH] hda: EAPD power management
by Matthew Ranostay 14 Apr '08

14 Apr '08

Power management support for EAPD enabled laptops, when headphones are sensed it pulls the EAPD GPIO line low to power it down. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 0dd2b6cb1506 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Wed Apr 09 08:16:33 2008 +0200 +++ b/pci/hda/patch_sigmatel.c Fri Apr 11 15:09:39 2008 -0400 @@ -129,6 +129,7 @@ struct sigmatel_spec { unsigned int hp_detect: 1; /* gpio lines */ + unsigned int eapd_mask; unsigned int gpio_mask; unsigned int gpio_dir; unsigned int gpio_data; @@ -3183,6 +3184,10 @@ static void stac92xx_hp_detect(struct hd for (i = 0; i < cfg->speaker_outs; i++) stac92xx_reset_pinctl(codec, cfg->speaker_pins[i], AC_PINCTL_OUT_EN); + if (spec->eapd_mask) + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, spec->gpio_data & + ~spec->eapd_mask); } else { /* enable lineouts, disable hp */ for (i = 0; i < cfg->line_outs; i++) @@ -3191,6 +3196,10 @@ static void stac92xx_hp_detect(struct hd for (i = 0; i < cfg->speaker_outs; i++) stac92xx_set_pinctl(codec, cfg->speaker_pins[i], AC_PINCTL_OUT_EN); + if (spec->eapd_mask) + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, spec->gpio_data | + spec->eapd_mask); } } @@ -3478,7 +3487,7 @@ again: spec->num_dmuxes = ARRAY_SIZE(stac92hd73xx_dmux_nids); spec->dinput_mux = &stac92hd73xx_dmux; /* GPIO0 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_dir = 0x1; + spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1; spec->gpio_data = 0x01; switch (spec->board_config) { @@ -3584,7 +3593,10 @@ again: spec->aloopback_shift = 0; /* GPIO0 High = EAPD */ - spec->gpio_mask = spec->gpio_dir = spec->gpio_data = 0x1; + spec->gpio_mask = 0x01; + spec->gpio_dir = 0x01; + spec->gpio_mask = 0x01; + spec->gpio_data = 0x01; spec->mux_nids = stac92hd71bxx_mux_nids; spec->adc_nids = stac92hd71bxx_adc_nids; @@ -3770,7 +3782,7 @@ static int patch_stac927x(struct hda_cod case STAC_D965_3ST: case STAC_D965_5ST: /* GPIO0 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_dir = 0x01; + spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x01; spec->gpio_data = 0x01; spec->num_dmics = 0; @@ -3787,7 +3799,7 @@ static int patch_stac927x(struct hda_cod /* fallthru */ case STAC_DELL_3ST: /* GPIO2 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_dir = 0x04; + spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x04; spec->gpio_data = 0x04; spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; @@ -3799,7 +3811,7 @@ static int patch_stac927x(struct hda_cod break; default: /* GPIO0 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_dir = 0x1; + spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1; spec->gpio_data = 0x01; spec->num_dmics = 0; @@ -3903,6 +3915,7 @@ static int patch_stac9205(struct hda_cod (AC_USRSP_EN | STAC_HP_EVENT)); spec->gpio_dir = 0x0b; + spec->eapd_mask = 0x01; spec->gpio_mask = 0x1b; spec->gpio_mute = 0x10; /* GPIO0 High = EAPD, GPIO1 Low = Headphone Mute, @@ -3912,7 +3925,7 @@ static int patch_stac9205(struct hda_cod break; default: /* GPIO0 High = EAPD */ - spec->gpio_mask = spec->gpio_dir = 0x1; + spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1; spec->gpio_data = 0x01; break; }

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[alsa-devel] [PATCH] hda: Correct SPDIF out default config
by Matthew Ranostay 14 Apr '08

14 Apr '08

Several laptops have have the SPDIF out defined as 'Digital other out' when it should be 'SPDIF out' in the default config. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 0dd2b6cb1506 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Wed Apr 09 08:16:33 2008 +0200 +++ b/pci/hda/patch_sigmatel.c Fri Apr 11 16:02:57 2008 -0400 @@ -3778,6 +3778,13 @@ static int patch_stac927x(struct hda_cod spec->mixer = stac927x_mixer; break; case STAC_DELL_BIOS: + switch (codec->subsystem_id) { + case 0x10280209: + case 0x1028022e: + /* correct the device field to SPDIF out */ + stac92xx_set_config_reg(codec, 0x21, 0x01442070); + break; + }; /* configure the analog microphone on some laptops */ stac92xx_set_config_reg(codec, 0x0c, 0x90a79130); /* correct the front output jack as a hp out */

2 1

[alsa-devel] [ALC262 patch_realtek.c] Fujitsu Lifebook PC speaker signal
by Tony Vroon 14 Apr '08

14 Apr '08

Good afternoon, Could you please consider the following patch for the 2.6.26 merge window. The legacy PC speaker signal was not routed to outputs. The codec is not prevented from powering down in this patch, although I suppose one could argue that perhaps it should be. Let me know if anyone feels strongly one way or the other. Signed-Off-By: Tony Vroon <tony(a)linx.net> Regards, Tony V. LINX sysadmin

2 1

[alsa-devel] [PATCH]PCI quirk for laptop LG which use CMI9880
by Jiang zhe 14 Apr '08

14 Apr '08

Hi, Please refer to [0003874] on the alsa mantis. This patch added the pci quirk. Thanks! Signed-off-by: Jiang zhe <zhe.jiang(a)intel.com>

2 1

[alsa-devel] [PATCH] soc - Include register in DAPM debug output
by Mark Brown 14 Apr '08

14 Apr '08

When logging register changes in DAPM debug output include the register number. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index abac684..e0637d8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -226,7 +226,7 @@ static int dapm_update_bits(struct snd_soc_dapm_widget *widget) snd_soc_write(codec, widget->reg, new); pop_wait(POP_TIME); } - dbg("reg old %x new %x change %d\n", old, new, change); + dbg("reg %x old %x new %x change %d\n", widget->reg, old, new, change); return change; } -- 1.5.4.4

2 1

[alsa-devel] [PATCH]PCI quirk for MSI laptop
by Jiang zhe 14 Apr '08

14 Apr '08

Hi, Please refer to [0003848] on the alsa mantis. This patch adds the pci quirk and Mic-Int controller. Thanks! Signed-off-by: Jiang zhe <zhe.jiang(a)intel.com>

2 1

[alsa-devel] [PATCH] snd_usb_caiaq: use likely()/unlikely()
by Daniel Mack 13 Apr '08

13 Apr '08

Use likely() and unlikely() in code which is called regularily. Signed-off-by: Daniel Mack <daniel(a)caiaq.de>

2 3

[alsa-devel] External IO plugin question
by Taek Joo Kim 12 Apr '08

12 Apr '08

Hi. I am working on ALSA external IO plugin now. Its function is similar to the pulse plugin. A server sends sound to multiple clients. When I play sound with my plugin, the sound is played really fast. It means that it is like playing with x5 or x10 speed. I did not write any code for synchronization between server and client. To try to fix it, I saw the pulse plugin. I guess that it uses some synchronization code (poll_descriptors), but I am not sure. My questions are: 1. Which part should take care of playing sound properly in this architecture, server or client? 2. Do I need to deal with synchronization between server and client? 3. If 2 is yes, is there any hints to do this with multiple clients? Thank you in advance.

1 0

[alsa-devel] Softvol for Independent Volume Control
by Cameron Barfield 11 Apr '08

11 Apr '08

Hi -- I've got an application that receives multiple network audio streams. Using DMIX, I am able to play back all the streams through my sound card. The problem is that I would like to be able to independently adjust the volume of each audio stream. Would Softvol work for this? All the examples I've found only show Softvol working as a replacement for a master volume control (and that's all I've been able to get it to do). If Softvol won't work for independent volume controls, is there any other ALSA plugin or API call that I'm missing? Switching to something like Pulseaudio would probably not be an option since I'm running in an embedded environment. Regards, Cameron

1 3

[alsa-devel] XRUN happens too often.
by Rong-Jhe 10 Apr '08

10 Apr '08

I use "default" pcm and open it in block mode. Then, I set the this pcm to perform 16-bit, two channels, and 44100 bps sampling rate audio data. The number of periods per buffer is 10. The follows are the parameters that I read from the pcm. sampling rate = 44100 bps period time = 21333 us period size = 940 frames buffer time = 213330 us buffer size = 9408 frames A sample in 16-bit, two channels is 4 bytes. I allocate a audio data of 9408x4=37632 bytes. audio_data=malloc(buffer_size*4); This audio_data buffer will receive audio data from the network. If this buffer is full, I use snd_pcm_writei( ) to put the buffer into the pcm. snd_pcm_sframes_t frames; snd_pcm_t *handle; while((frame=snd_pcm_writei(handle, audio_data, buffer_size))<0) { if(frames==-EPIPE) snd_pcm_prepare(handle); else printf("snd_pcm_writei error %d\n", frames); } The codes states that I put the entire buffer into the pcm at each time. The pcm use 10 periods to perform this buffer. The program can work normally. If the sampling rate is changed, the program will encounter XRUN frequently. The follows are the parameters that I read from the pcm after setting sampling rate to 48000. sampling rate = 48000 bps period time = 21333 us period size = 1024 frames buffer time = 213330 us buffer size = 10240 frames audio_data=malloc(buffer_size*4); //10240x4=40960 I also put the entire buffer into the pcm at each time, but snd_pcm_writei( ) will return -PIPE frequently. What is the problem?

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[alsa-devel] M-Audio Fast Track Ultra USB
by knoxit.etc＠gmail.com 10 Apr '08

10 Apr '08

Hello, I was wondering if any user had experiences with the M-Audio Fast Track Ultra USB unit and ALSA. I'm not sure if its supported, it doesn't show up on the soundcard matrix. If not, is there any development taking place right now? Thanks.

2 1

[alsa-devel] [PATCH] snd_usb_caiaq: correct input channel order
by Daniel Mack 09 Apr '08

09 Apr '08

This patch corrects the input channel order of hardware supported by snd_usb_caiaq. Signed-off-by: Daniel Mack <daniel(a)caiaq.de>

1 0

[alsa-devel] HDSP maintainer
by Florian Faber 07 Apr '08

07 Apr '08

Hi there, is there a current maintainer for the RME drivers? I have several patches and additions for the hdsp driver and the user space applications, and there will be much more coming. Flo -- Machines can do the work, so people have time to think. public key 6C002249 x-hkp://wwwkeys.eu.pgp.net

1 0

[alsa-devel] Sound card question.
by Tobin Davis 07 Apr '08

07 Apr '08

Are there any alsa supported sound cards that do hardware audio decoding for MP3/WMA and otehr formats? I'd like to look at the mechanism used to connect the cards to the data stream. Also, is there any block diagrams I can look at for the AOA and SOC drivers? Particularly the I2S bus support. I'm ramping up support for a new project that will be Alsa based, and would like to combine it with already established APIs if possible. -- Tobin Davis The relative importance of files depends on their cost in terms of the human effort needed to regenerate them. -- T.A. Dolotta

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[alsa-devel] which fonction ?
by Dsant 07 Apr '08

07 Apr '08

I would like to implement this, but I don't know which fonction to use : LOOP START begin acquire some long stuf... (about 1-2 seconds) copy buffer to ram or into file LOOP END Dsant

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[alsa-devel] Is there a list of alsa error codes?
by Wolfgang Woehl 06 Apr '08

06 Apr '08

Hi, For the life of me I can't find a list of alsa error codes in order to understand what, in this case, "sound/usb/usbaudio.c:352: frame 0 active: -18" means. A hint, anyone? Wolfgang

1 0

[alsa-devel] Loop sequencer
by Guillaume Dupré 06 Apr '08

06 Apr '08

Hello I try to program a loop midi sequencer with libasound . For the moment, I have this : #include <alsa/asoundlib.h> #include <unistd.h> using namespace std; int main() { int aa; snd_seq_t* seq; int port,port2, queue; snd_seq_queue_tempo_t* tempo; snd_seq_event_t ev; snd_seq_open(&seq, "default", SND_SEQ_OPEN_DUPLEX, 0); snd_seq_set_client_name(seq, "monprog"); port = snd_seq_create_simple_port(seq, "port1",SND_SEQ_PORT_CAP_READ | SND_SEQ_PORT_CAP_SUBS_READ,SND_SEQ_PORT_TYPE_APPLICATION); queue = snd_seq_alloc_queue(seq); //TEMPO snd_seq_queue_tempo_alloca(&tempo); snd_seq_queue_tempo_set_tempo(tempo, 60000000 / 120); snd_seq_queue_tempo_set_ppq(tempo, 4); snd_seq_set_queue_tempo(seq, queue, tempo); snd_seq_connect_to(seq, port, 130, 0); snd_seq_start_queue(seq, queue, NULL); // write 2 notes snd_seq_ev_clear(&ev); snd_seq_ev_set_subs(&ev); snd_seq_ev_set_source(&ev, port); snd_seq_ev_schedule_tick(&ev, queue, 0, 0); snd_seq_ev_set_note(&ev, 0, 60, 127, 3); snd_seq_event_output(seq, &ev); ev.time.tick = 4; ev.data.note.note = 67; snd_seq_event_output(seq, &ev); snd_seq_drain_output(seq); sleep(4) ; return 0; } It works, no problem . But now, I want to do a loop with my two notes . I searched on the Internet but I didn't find . Thanks Guillaume Dupré

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Re: [alsa-devel] 2.6.25-rc8-mm1 -- INFO: possible circular locking dependency detected (while using iw to debug a wireless issue)
by Johannes Berg 04 Apr '08

04 Apr '08

On Thu, 2008-04-03 at 10:41 -0400, Miles Lane wrote: > After collecting the wireshark log, I found this in my messages log: Below is a better version of the trace Miles sent me privately, but I have no idea what would be causing it. I don't think it's related to wireless. ======================================================= [ INFO: possible circular locking dependency detected ] 2.6.25-rc8-mm1 #15 ------------------------------------------------------- iw/9417 is trying to acquire lock: (genl_mutex){--..}, at: [ctrl_dumpfamily+0x37/0xda] ctrl_dumpfamily+0x37/0xda but task is already holding lock: (nlk->cb_mutex){--..}, at: [netlink_dump+0x39/0x22e] netlink_dump+0x39/0x22e which lock already depends on the new lock. the existing dependency chain (in reverse order) is: -> #1 (nlk->cb_mutex){--..}: [__lock_acquire+0xa02/0xbaf] __lock_acquire+0xa02/0xbaf [lock_acquire+0x76/0x9d] lock_acquire+0x76/0x9d [snd_mixer_oss:mutex_lock_nested+0xd5/0x67b] mutex_lock_nested+0xd5/0x274 [netlink_dump_start+0xbd/0x15a] netlink_dump_start+0xbd/0x15a [genl_rcv_msg+0x9d/0x13a] genl_rcv_msg+0x9d/0x13a [netlink_rcv_skb+0x30/0x75] netlink_rcv_skb+0x30/0x75 [genl_rcv+0x1e/0x2e] genl_rcv+0x1e/0x2e [cfg80211:netlink_unicast+0x1c1/0x1867] netlink_unicast+0x1c1/0x293 [netlink_sendmsg+0x21f/0x22c] netlink_sendmsg+0x21f/0x22c [sock_sendmsg+0xca/0xe1] sock_sendmsg+0xca/0xe1 [sys_sendmsg+0x14d/0x1a8] sys_sendmsg+0x14d/0x1a8 [sys_socketcall+0x163/0x17e] sys_socketcall+0x163/0x17e [sysenter_past_esp+0x6d/0xc5] sysenter_past_esp+0x6d/0xc5 [<ffffffff>] 0xffffffff -> #0 (genl_mutex){--..}: [__lock_acquire+0x929/0xbaf] __lock_acquire+0x929/0xbaf [lock_acquire+0x76/0x9d] lock_acquire+0x76/0x9d [snd_mixer_oss:mutex_lock_nested+0xd5/0x67b] mutex_lock_nested+0xd5/0x274 [ctrl_dumpfamily+0x37/0xda] ctrl_dumpfamily+0x37/0xda [netlink_dump+0x51/0x22e] netlink_dump+0x51/0x22e [netlink_recvmsg+0x156/0x203] netlink_recvmsg+0x156/0x203 [sock_recvmsg+0xd1/0xe9] sock_recvmsg+0xd1/0xe9 [sys_recvmsg+0xf2/0x17f] sys_recvmsg+0xf2/0x17f [sys_socketcall+0x16f/0x17e] sys_socketcall+0x16f/0x17e [sysenter_past_esp+0x6d/0xc5] sysenter_past_esp+0x6d/0xc5 [<ffffffff>] 0xffffffff other info that might help us debug this: 1 lock held by iw/9417: #0: (nlk->cb_mutex){--..}, at: [netlink_dump+0x39/0x22e] netlink_dump+0x39/0x22e stack backtrace: Pid: 9417, comm: iw Not tainted 2.6.25-rc8-mm1 #15 [print_circular_bug_tail+0x5b/0x66] print_circular_bug_tail+0x5b/0x66 [print_circular_bug_header+0xa5/0xb0] ? print_circular_bug_header+0xa5/0xb0 [__lock_acquire+0x929/0xbaf] __lock_acquire+0x929/0xbaf [lock_acquire+0x76/0x9d] lock_acquire+0x76/0x9d [ctrl_dumpfamily+0x37/0xda] ? ctrl_dumpfamily+0x37/0xda [snd_mixer_oss:mutex_lock_nested+0xd5/0x67b] mutex_lock_nested+0xd5/0x274 [ctrl_dumpfamily+0x37/0xda] ? ctrl_dumpfamily+0x37/0xda [ctrl_dumpfamily+0x37/0xda] ? ctrl_dumpfamily+0x37/0xda [ctrl_dumpfamily+0x37/0xda] ctrl_dumpfamily+0x37/0xda [netlink_dump+0x51/0x22e] netlink_dump+0x51/0x22e [mac80211:kfree_skb+0x40/0x45] ? kfree_skb+0x40/0x45 [netlink_recvmsg+0x156/0x203] netlink_recvmsg+0x156/0x203 [sock_sendmsg+0xca/0xe1] ? sock_sendmsg+0xca/0xe1 [sock_recvmsg+0xd1/0xe9] sock_recvmsg+0xd1/0xe9 [<c01313d9>] ? autoremove_wake_function+0x0/0x30 [snd_pcm_oss:copy_from_user+0x3b/0x236] ? copy_from_user+0x3b/0x5e [verify_iovec+0x40/0x70] ? verify_iovec+0x40/0x70 [sys_recvmsg+0xf2/0x17f] sys_recvmsg+0xf2/0x17f [snd_hda_intel:_spin_unlock_irqrestore+0x56/0x6c] ? _spin_unlock_irqrestore+0x56/0x6c [paravirt_get_lazy_mode+0xe/0x1b] ? paravirt_get_lazy_mode+0xe/0x1b [crypto_algapi:kunmap_atomic+0x9e/0x2e8e] ? kunmap_atomic+0x9e/0xbb [handle_mm_fault+0x7aa/0x7bb] ? handle_mm_fault+0x7aa/0x7bb [sys_socketcall+0x16f/0x17e] sys_socketcall+0x16f/0x17e [processor:trace_hardirqs_on+0xf0/0x5d38] ? trace_hardirqs_on+0xf0/0x12c [sysenter_past_esp+0x6d/0xc5] sysenter_past_esp+0x6d/0xc5

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[alsa-devel] Edirol M-16 DX
by James Trevelyan 04 Apr '08

04 Apr '08

I recently bought an Edirol M-16DX and am trying to get it working with Linux. I'm almost there but would appreciate some help in getting it over the line. The M-16DX (http://www.rolandus.com/products/productdetails.aspx?ObjectId=860&ParentId= 114) is a fairly new and pretty neat piece of kit - it works as a standalone 16-channel hardware mixer, with a USB2.0 connection which can send each separate input to the computer, with the ability also to send a stereo stream back to the mixer. In addition it can be put in a DAW controller mode where it acts a control surface with almost all the mixer knobs usable. It's set up as a vendor specific device so at first nothing happens when the usb cable connected. The usb configuration has three interfaces, like some of the other Edirol devices. So I put a new quirk in usbquirks.h for the appropriate device id, with a QUIRK_AUDIO_STANDARD_INTERFACE on interfaces 1 and 2, and a QUIRK_MIDI_FIXED_ENDPOINT on interface 3. Accodring to lsusb -v, each interface provides what looks like a proper class-specific descriptor. Having recompiled, the system now detects the device and loads the usb-audio module when the device is plugged in. It successfully creates alsa devices for capture and playback, as well as midi, with the correct sample rate (44.1, 48 or 96 as set on the device), correct number of channels (18 in and 2 out). All looks as it should. When I aplay an audio file to the device, it plays fine - almost. Everything sounds good, except that every 9-10 seconds, there is a kind of nasty high pitched digital distortion which lasts for a second. (definitely not in the audio file!) This seems to be synchronised to the start of the stream - when I aplay a file, it opens a new stream and the distortion is always in the same places in the song. If I open the ports on jack so the stream stays open, and play music through it on ardour, the distortion is at different points in the song (always about 9-10 seconds apart). I haven't tried capture yet. It all runs fine under windows in the same configuration. I'm at a bit of a loss as to the problem here - perhaps there is something wrong with the detection, and I haven't yet tried creating an AUDIO_FIXED_ENDPOINT quirk to play with the settings manually. But I would have thought in that case it just wouldn't work at all. I know the sample rate and # channels is correct. My system's pretty up to date (2.6.24.3 kernel, alsa-libs 1.0.16). I can provide full lsusb and /proc/asound information as necessary. Suggestions welcome! James

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[alsa-devel] Altec Lansing FX5051 mixer
by Jeremy Nickurak 04 Apr '08

04 Apr '08

I recently purchased an Altec Lansing USB Surround speaker set. (It's a set of 5.1 speakers, with a built-in USB sound card). I've got about a week before the return policy expires and I'd have to return it. The mixer doesn't work at all w/linux kernel 2.6.22, everything is locked at 100% volume, which is pretty unhelpful, as it's the only way to adjust the volume. (no analog controls). Also, the digital control panel locks up when the speakers power up, further preventing any access to the volume controls. The device works out-of-the-box without any drivers under XP, so I have to so-far assume that alsa's usb-audio is missing something. Any thoughts as to what could be wrong? I'm more than happy to prod the device and submit any information I can extract about it, or even potentially opening the device up for remote access if a reputable developer is interested in examining it. -- Jeremy Nickurak -= Email/Jabber: atrus(a)rifetech.com =-

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Re: [alsa-devel] + es1968-fix-jitter-on-some-maestro-cards.patch added to -mm tree
by Rene Herman 03 Apr '08

03 Apr '08

On 03-04-08 20:54, akpm(a)linux-foundation.org wrote: > The patch titled > es1968: fix jitter on some maestro cards > has been added to the -mm tree. Its filename is > es1968-fix-jitter-on-some-maestro-cards.patch [ ... ] > Subject: es1968: fix jitter on some maestro cards > From: Andreas Mueller <andreas(a)stapelspeicher.org> > > This patch suppresses jitter on several Maestro cards in stereo mode (ALSA of > course). > > The patch is also incorporated in the *BSD drivers where I "ported" it from. > > Without this patch most of the stereo audio gets out of sync and really > distorted (oss-emulation with mplayer at 48000khz worked somehow). > > Cc: Rene Herman <rene.herman(a)keyaccess.nl> > Cc: Jaroslav Kysela <perex(a)suse.cz> > Cc: Takashi Iwai <tiwai(a)suse.de> > Signed-off-by: Andrew Morton <akpm(a)linux-foundation.org> > --- Tested-By: Rene Herman <rene.herman(a)gmail.com> on a TerraTec DMX (ESS Canyon3D, ES1970MS-3D) which doesn't need this but doesn't seem to be suffering from it either. No idea about this otherwise. > sound/pci/es1968.c | 22 +++++++++++++++++++++- > 1 file changed, 21 insertions(+), 1 deletion(-) > > diff -puN sound/pci/es1968.c~es1968-fix-jitter-on-some-maestro-cards sound/pci/es1968.c > --- a/sound/pci/es1968.c~es1968-fix-jitter-on-some-maestro-cards > +++ a/sound/pci/es1968.c > @@ -1815,6 +1815,23 @@ snd_es1968_pcm(struct es1968 *chip, int > > return 0; > } > +/* > + * suppress jitter on some maestros when playing stereo > + */ > +static void snd_es1968_suppress_jitter(struct es1968 *chip, struct esschan *es) > +{ > + unsigned int cp1; > + unsigned int cp2; > + unsigned int diff; > + > + cp1 = __apu_get_register(chip, 0, 5); > + cp2 = __apu_get_register(chip, 1, 5); > + diff = (cp1 > cp2 ? cp1 - cp2 : cp2 - cp1); > + > + if (diff > 1) { > + __maestro_write(chip, IDR0_DATA_PORT, cp1); > + } > +} > > /* > * update pointer > @@ -1936,8 +1953,11 @@ static irqreturn_t snd_es1968_interrupt( > struct esschan *es; > spin_lock(&chip->substream_lock); > list_for_each_entry(es, &chip->substream_list, list) { > - if (es->running) > + if (es->running) { > snd_es1968_update_pcm(chip, es); > + if (es->fmt & ESS_FMT_STEREO) > + snd_es1968_suppress_jitter(chip, es); > + } > } > spin_unlock(&chip->substream_lock); > if (chip->in_measurement) { Rene.

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[alsa-devel] HDA-Intel ALC880, fujitsu siemens amilo PI1536 and flavours
by Luca Bonora 03 Apr '08

03 Apr '08

Hello developers. I've got a problem with my laptop, I've asked help on alsa-user and now I have a request for you (if it's possible). For a year I tried to use my microphone unsuccessfully. Now I've found a solution, that is to use the flavour "6stack"/"6stack-digout" in place of the flavour "fujitsu". I've read the ALSA-configuration.txt and fujitsu is properly the flavour for _my_ laptop but with 6stack-digout the microphone works and the volume control does not work. Is it possible to 'merge' the two flavours to have both microphone and volume control working? Can I ask it to you? Thank you for your help! Luca

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[alsa-devel] [PATCH] snd_usb_caiaq: fix potential lockups locking
by Daniel Mack 03 Apr '08

03 Apr '08

Hi, attached is a patch for snd_usb_caiaq which fixes system potential lockups when stoping and restarting the stream very quickly multiple times in a row which is what some players do when the user drags the song position slider. Also, it seems that snd_pcm_period_elapsed() must not be called when holding a spinlock? The documentation[*], however, doesn't state that. As this fixes an annoying bug, I would appreciate seeing that one being pushed into 2.6.25. Still possible? Thanks and best regards, Daniel [*] http://www.alsa-project.org/~tiwai/writing-an-alsa-driver/x773.htm#PCM-INTE… ---- Subject: [PATCH] snd_usb_caiaq: fix potential lockups This patch fixes potential lockups in snd_usb_caiaq by refining the locking mechanims and by using usb_kill_urb() in favor to usb_unlink_urb(). Signed-off-by: Daniel Mack <daniel(a)caiaq.de>

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[alsa-devel] Problems in writing test apps
by haoshun 03 Apr '08

03 Apr '08

Hi: sorry,forget to add the subject right now... I'm confused by a problem when i modify a test application "alsa-lib-1.0.13/test/control.c". I wanna use the structure "snd_ctl_elem_value_t",but when i write like this: snd_ctl_elem_value_t aaa; there will be an error: storage size of 'aaa' isn't known while snd_ctl_elem_value_t *aaa; makes no errors. But i can't change it directly such as: aaa->id=test_id ; //test_id is defined before cause it will bring an error: dereferencing pointer to incomplete type I think i should use the function like "snd_ctl_elem_info_set_id",but i just want to know why,i don't think it disobeys the C syntax.Is there something different in the ALSA making procedure?or Makefile? haoshun 2008-04-03

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[alsa-devel] (no subject)
by 郝舜 03 Apr '08

03 Apr '08

Hi: I'm confused by a problem when i modify a test application "alsa-lib-1.0.13/test/control.c". I wanna use the structure "snd_ctl_elem_value_t",but when i write like this: snd_ctl_elem_value_t aaa; there will be an error: storage size of 'aaa' isn't known while snd_ctl_elem_value_t *aaa; makes no errors. But i can't change it directly such as: aaa->id=test_id ; //test_id is defined before cause it will bring an error: dereferencing pointer to incomplete type I think i should use the function like "snd_ctl_elem_info_set_id",but i just want to know why,i don't think it disobeys the C syntax.Is there something different in the ALSA making procedure?or Makefile? haoshun 2008-04-03

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[alsa-devel] ASoC for TI OMAP - Multiformat ASoC DAI link driver?
by Jarkko Nikula 03 Apr '08

03 Apr '08

Currently in ASoC v1 we have one DAI driver per one format? E.g. separate drivers for I2S, AC97, etc and type field in cpu_dai structure is not a bit field or at least not used such way in soc-core.c and DAI link drivers. struct snd_soc_cpu_dai pxa_i2s_dai = { ... .type = SND_SOC_DAI_I2S, ... I looked shortly asoc-v2-dev branch and now this type field is gone. Does it mean that there you can have DAI link driver which supports multiple formats in set_fmt callback? I'm asking this since McBSP (Multichannel Buffered Serial Port) found in OMAP processors can be configured to almost any imaginable format and IMO it doesn't make a sense to have separate drivers for each of them if there is a difference only in few configuration bits. -- Jarkko

2 2

[alsa-devel] ASoC for TI OMAP - alsa-devel submission?
by Jarkko Nikula 03 Apr '08

03 Apr '08

I've been doing generic ASoC drivers for Texas Instruments OMAP processors and I recently released the first version on linux-omap mailing list: http://marc.info/?l=linux-omap&m=120643770810683&w=2 http://marc.info/?l=linux-omap&m=120643766610623&w=2 Currently patch tries to cover OMAPs from 1x10 to 2420 and is tested on Nokia N810 having OMAP2420 and TLV320AIC33 codec. I'm going to release it here also but there are few issues I would like to discuss before it: - McBSP used in OMAP2430 and OMAP3430 are bit different than other 1510-2420 OMAPs and that support we have currently missing in linux-omap either. However, changes to my drivers should be quite minimal in order to support them. E.g. base address definitions and different DMA data type. - Solving the channel synchronization problem might require API change to linux-omap McBSP driver. - ASoC v2 considerations? Like multiformat DAI. - Multiformat McBSP DAI link driver (I'll send an another mail on that). So what I mean that should I post the patch like as it now, rebased to dev branch of linux-2.6-asoc tree and with some minor changes, or wait a bit when all linux-omap and ASoC v2 evolve? What you think? -- Jarkko

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[alsa-devel] DMIX and Alsa Drivers
by Barclay, Bruce 03 Apr '08

03 Apr '08

I wrote an Alsa driver with copy callbacks and it worked. Later read that you can use copy callbacks if you want to use dmix so I rewrote it to not use copy callbacks. I don't see much documentation on dmix (/etc/asound.conf). When I put dmix in there, I can run two versions but the framework never updates the application pointer so I get no data. I take the dmix out and it works fine. No clue on what to do. Bruce

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Re: [alsa-devel] PXA2xx/3xx AC97 Driver(s)
by Mark Brown 02 Apr '08

02 Apr '08

On Mon, Mar 24, 2008 at 12:38:00PM +0000, Oliver Ford wrote: > Is there any work currently under way on an AC97 driver for the PXA3xx? There is PXA3xx AC97 support present in the ASoC tree: git://opensource.wolfsonmicro.com/linux-2.6-asoc It's under the asoc-v2-dev branch at the minute and is probably only done for the ASoC AC97 driver, I'd need to check. The main reason it's only there is that it depends on the ARM architecture changes that you identified (which are also applied on the asoc-v2-dev branch) and it makes life easier to manage the cross-tree changes in only one tree. All the ARM architecture changes have been submitted and marked as applied in the patch system so should hopefully appear in Linus' kernel early in the 2.6.26 cycle. Once this has happened I will submit the AC97 driver PXA3xx support. > I want to get the touch screen working on my iPAQ 214, which uses the > PXA310. It's a WM97xx chip on the AC97 interface. I'm not really too I know that there are a number of users currently using the tree for that use case so the asoc-v2-dev branch should hopefully work for you. If you run into trouble please let me know. > include/asm-arm/arch-pxa/pxa3xx-regs.h). Does the pxa310 require > anything special like this or should just setting the cold-reset in > the AC97 GCR work? It also doesn't support the use of interrupts during reset operations, requiring another special case, but I think that's about it. > The Wolfson Microelectronics open-source page indicates that work is on > going with the 3xx driver but there doesn't seem to be anything more > than the pxa2xx driver in their source tree. Does anyone know how active > / close to producing anything this is? Like I say, it's there. The changes are fairly straightforward and should be easy to backport to whatever kernel you need - if you need help identifying what needs merging over please just ask. > More generally, I'm a little confused as to why there is both the > standard (sound/arm/pxa2xx-ac97.c) and SoC (sound/soc/pxa/pxa2xx-ac97) > driver. Is one of these obsolete now or are they both still in > development? If guessing, I'd say the SoC driver is now the one to use The SoC driver is generally preferable, though it requires a bit more work to use (since you have to write a platform driver to use it). Both are still maintained. SoC offers better power management and support for a wider range of board configurations, especially those that also have non-AC97 interfaces (such as the I2S interface on the WM97xx parts). > but am not sure. Compiling either gives the warning "'pxa_set_cken' is > deprecated". I suspect calling this won't work for the pxa310 since it's This is also fixed in the patches currently queued in the ARM tree - the drivers have been converted to use the clk_() API. > clocks are different, but given that I'm booting linux from windows > (which will leave the clock on) and don't particularly care about > suspend/resume, should not being able to switch it off be a problem? The cken calls should actually work fine.

2 3

[alsa-devel] Marian sound card support
by Peter Brooks 02 Apr '08

02 Apr '08

Hi all, I'm looking at a Marian 8 sound card (http://www.marian.de/en/products/trace8), and it has no mention on the alsa site. Is there any support for this card? I'm looking for a sound card with the maximum amount of outputs, I don't need many inputs. Currently the only one after this card I see is an M-Audio 1010. Cheers -- Peter Brooks

1 0

[alsa-devel] snd-powermac
by Risto Suominen 02 Apr '08

02 Apr '08

I'm about to send the following to alsa-user mailing list: I've been trying to fix ALSA support for older PowerMacs. Anyone experiencing problems - like missing or inoperative or wrongly directed controls in alsamixer - could you, please, test my patch and write the results into this list or directly to me? If you are still with me, check first with lsmod that you are using snd-powermac: $ lsmod | grep ^snd.powermac snd_powermac 56576 0 $ Then you will need to install make, C-compiler and kernel headers corresponding to your running kernel, f.ex (Ubuntu Dapper 6.06.1): $ sudo aptitude install make gcc linux-headers-2.6.15-26-powerpc and fetch the ALSA driver source code and my patch: $ wget ftp://ftp.alsa-project.org/pub/driver/alsa-driver-1.0.16.tar.bz2 $ wget http://ristosu.wippiespace.com/pub/alsa-driver-1.0.16-p4.diff Unpack the source and apply the patch: $ tar xjf alsa-driver-1.0.16.tar.bz2 $ cd alsa-driver-1.0.16 $ patch -p0 < ../alsa-driver-1.0.16-p4.diff Finally configure: $ ./configure --disable-verbose-printk --with-kernel=/usr/src/linux-headers-2.6.15-26-powerpc --with-build=/usr/src/linux-headers-2.6.15-26-powerpc --with-moddir=/lib/modules/2.6.15-26-powerpc/updates/sound --with-isapnp=no --with-oss=yes --with-pcm-oss-plugins=yes --with-cards=powermac make and install: $ make $ sudo make install-modules To load the new modules without rebooting: $ sudo modprobe -r snd-powermac $ sudo modprobe -r snd-pcm-oss $ sudo modprobe snd-powermac Risto

1 3

[alsa-devel] Alsa : GPL or LGPL ?
by liste＠votreservice.com 02 Apr '08

02 Apr '08

Wikipedia says Alsa is under the GPL and LGPL ? Can I use Alsa for recording and playing sound in my proprietary game under Linux ? Dsant

2 1

[alsa-devel] Detecting new alsa sequencer clients (with some event loop) so that i can subscribe them
by R. Dresens 02 Apr '08

02 Apr '08

Hello, I'm currently writing an application that will be able to 'auto subscribe' itself to a certain sequencer client when that client becomes available to the system. I want (for example) to be able to plug in a USB MIDI interface, and when I do that then my application detects this and does a subscription on it, just like aconnect would do. A possible solution is polling; every second or so, i could generate a list of clients with snd_seq_query_next_client and look for a certain client name (libpcre comes to mind here;), but I was wondering if there is a better solution? ..in other words, is it possible to get such events from the system in a similar way like one would receive regular MIDI events? http://www.alsa-project.org/alsa-doc/alsa-lib/structsnd__seq__event.html list all the events; an interesting one could be snd_seq_connect_t for monitoring connections on my own ports, but that's not what I'm looking for. I can't seem to find an event related to this functionality. Is there something what I'm currently overlooking? Such functionality may be usefull for GUI versions of aconnect. I looked in the source of qjackctl (it has an alsa patchbay), but I haven't found nothing usefull yet since it seems to me that this software also does polling/refresh on a certain interval. Has anyone a usefull tip for me? Thanks in advance, Greetings, Raymond Dresens.

2 2

[alsa-devel] [Fwd: Re: [regression] 2.6.25-rc4 snd-es18xx broken on Alpha]
by Tyson Whitehead 02 Apr '08

02 Apr '08

Sorry. Bob was just saying this could be an important observation, but it didn't get to everyone as the CC list got shortened. Cheers! -Tyson

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Re: [alsa-devel] [regression] 2.6.25-rc4 snd-es18xx broken on Alpha
by rct＠frus.com 02 Apr '08

02 Apr '08

rct wrote: > Rene Herman wrote: > > Okay, and applying the attached just makes your sound completely dead in the > > water? > > (patch to remove es1888_init from a Miata build omitted) > > I'll try a build with the old OSS "sb" driver, and if that works ok, we > may be able to do away with es1888_init() on the Miata. Tyson -- I > think you have a Miata if I'm remembering correctly: can you confirm > these observations? Quick followup: OSS "sb" driver works fine without es1888_init(). -- ------------------------------------------------------------------------ Bob Tracy | "I was a beta tester for dirt. They never did rct(a)frus.com | get all the bugs out." - Steve McGrew on /. ------------------------------------------------------------------------

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[alsa-devel] How to reduce delay between playback and capture for echo canceller
by Cliff Cai 01 Apr '08

01 Apr '08

Hi all, I'm playing a linphone based VOIP demo board these days,the board can connect different soundcard daughter board as playback and capture devices through I2S or AC97.Then I experience a bad echo on my PC(using liphone and headset) when talk with the board (with speaker and MIC on board).So I enable the echo canceller (based on speex) for it. One of my sound card (connected by I2S) works pretty well,but the other one (connected by AC97) doesn't work. Through the diagnostic tool provided by speex I find that the ac97 one has a big delay between played signal(reference signal) and captured signal.Just the delay make echo canceller don't work as expectation.So I think the driver for the AC97 soundcard need some improvement to reduce the delay. Has anyone a usefull direction for me? Thanks in advance,Best Regards Cliff Cai _________________________________________________________________ MSN 中文网，最新时尚生活资讯，白领聚集门户。 http://cn.msn.com

1 0

[alsa-devel] [Fwd: Re: Davinci ASoC support (patch)]
by lokowich 01 Apr '08

01 Apr '08

Forwarded from davinci-linux-open-source list: I've built this ASoC patch into my DaVinci git kernel (thanks for the SENSORS tip), and can use aplay and speaker-test, but no sound. Figured it may be a mixer/volume control issue, but the alsamixer is unable to open for a failed ioctl on USBDEVFS_IOCTL. Not sure why there is a USB dependency. Noted that there is no /sys/class/sound/mixer in my sysfs. Any ideas? Mark Bernard Blackham wrote: > Behme wrote: > >> While searching something different, I just found >> >> http://www.spinics.net/lists/alsa-devel/msg12535.html >> >> As I'm not subscribed to alsa-devel, I don't have the original post. >> Maybe any of the authors can post it do this list as well? >> >> The patch mainly applies to our git. I fixed two small rejects in >> Kconfig and Makefile as the patch is for newer kernel than our git. >> Untested patch against our git in attachment. Anybody likes to test? >> > > Well, I tried compiling it and it failed to build. It appears that it's > built on a set of patches for Davinci posted to LAK, including this > completely revamped EDMA support[1]: > > http://www.spinics.net/lists/arm-kernel/msg47450.html > > So I had to change a few definitions, import a few functions from the new > DMA support and managed to get it compiling and loading, but not yet working > (tried various sample rates and formats): > > root@davinci-dvevm:/tst$ aplay test.wav > Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo > ALSA lib pcm_params.c:2152:(snd_pcm_hw_refine_slave) Slave PCM not usable > aplay: set_params:878: Broken configuration for this PCM: no configurations > available > > root@davinci-dvevm:/tst$ arecord -r 48000 -f S16_LE > Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono > ALSA lib pcm_params.c:2152:(snd_pcm_hw_refine_slave) Slave PCM not usable > arecord: set_params:878: Broken configuration for this PCM: no > configurations available > > > And this is the extent of my knowledge of ALSA - I can continue digging, but > maybe somebody else will beat me to it =) It's looking promising though. > > Attached is the patch I have so far that applies against current git. This > is not for going upstream, but just for testing. Notes: > - Apply patch > - Enable SND_SOC, SND_DAVINCI_SOC, SND_DAVINCI_SOC_I2S, SND_DAVINCI_SOC_EVM > and SND_SOC_TLV320AIC3X. > - Make sure CONFIG_I2C_DAVINCI is enabled, but disable the > CONFIG_SENSORS_TLV320* options. > - Build. > - Load the snd-soc-evm module. > > Bernard. > > [1] These patches appear to have gone absolutely nowhere beyond that list. I > can't find them in any tree. But they are cleaner and more conforming to the > Linux Kernel style than what's currently in git. > > > ------------------------------------------------------------------------ > > _______________________________________________ > Davinci-linux-open-source mailing list > Davinci-linux-open-source(a)linux.davincidsp.com > http://linux.davincidsp.com/mailman/listinfo/davinci-linux-open-source >

4 8

[alsa-devel] Keep getting underruns
by Barclay, Bruce 31 Mar '08

31 Mar '08

Not sure what causes this in the Alsa framework. I only get one at the beginning of the song but not sure how and why I get it. Does it have to do with the return of my pointer callback or because I call snd_period_elapsed in my interrupt routine too soon. Any ideas or docs to read? Bruce

1 0

[alsa-devel] M-Audio Revolution 7.1 sound card causing crash on Alpha
by Michael Cree 31 Mar '08

31 Mar '08

I recently installed a M-Audio Revolution 7.1 sound card into DEC/Compaq Alpha XP1000, and it is causing crashes and complete lock ups of the computer when I attempt to play sound through the sound card. Since this is my main computer I shifted the card to a DEC Alpha PWS500au and it causes the same lock ups there, and in one case a kernel oops made it into the system log (listed below). The crashes occur when any application plays sound through the card. The first instance of running an application (such as aplay or speaker-test) after bootup succeeds; it's the second time one runs such an application that the system crashes. I am using the ice1724 alsa driver. The sound card works fine - I've verified it on a MS Windows machine. It also worked fine when I installed it in a x86_64 machine running Debian testing. This seems to be a bug possibly specific to the Alpha architecture. Any ideas as to the problem? Any suggestions as to effective debugging? I turned on the debug options for alsa in the kernel config and re-compiled, but couldn't find any alsa debugging information in any of the system logs. Basic system details follow: DEC Alpha PWS500au with a 600MHz Alpha EV56 cpu. Debian GNU/Linux - up to date testing (Lenny) distribution. 2.6.24.3 verbatim kernel self-compiled and alsa as provided in that kernel. The kernel oops from one crash is as follows: Mar 4 22:04:04 aleph kernel: Kernel bug at mm/mmap.c:2054 Mar 4 22:04:04 aleph kernel: speaker-test(1885): Kernel Bug 1 Mar 4 22:04:04 aleph kernel: pc = [<fffffc000036bec4>] ra = [<fffffc000036be88>] ps = 0000 Not tainted Mar 4 22:04:04 aleph kernel: pc is at exit_mmap+0x134/0x150 Mar 4 22:04:04 aleph kernel: ra is at exit_mmap+0xf8/0x150 Mar 4 22:04:04 aleph kernel: v0 = 0000000000000000 t0 = 0000000000000003 t1 = 0000000000000040 Mar 4 22:04:04 aleph kernel: t2 = 000000000000003f t3 = fffffc002300c5f8 t4 = 0000000000000001 Mar 4 22:04:04 aleph kernel: t5 = fffffc00001da000 t6 = 0000000000000000 t7 = fffffc00212dc000 Mar 4 22:04:04 aleph kernel: a0 = 0000000000000000 a1 = fffffc002300c400 a2 = 0000000000000000 Mar 4 22:04:04 aleph kernel: a3 = 0000000000000000 a4 = 0000000000000000 a5 = 0000000000000000 Mar 4 22:04:04 aleph kernel: t8 = 0000000000000000 t9 = 0000000d1bc1e51d t10= 1a00000000000000 Mar 4 22:04:04 aleph kernel: t11= 0000000000002000 pv = fffffc000037b3d0 at = 0000000000000003 Mar 4 22:04:04 aleph kernel: gp = fffffc00006fb518 sp = fffffc00212dfbe8 Mar 4 22:04:04 aleph kernel: Trace: Mar 4 22:04:04 aleph kernel: [<fffffc000032516c>] mmput+0x5c/0x100 Mar 4 22:04:04 aleph kernel: [<fffffc0000329820>] exit_mm+0xc0/0x180 Mar 4 22:04:04 aleph kernel: [<fffffc000032a95c>] do_exit+0x16c/0x950 Mar 4 22:04:04 aleph kernel: [<fffffc000032b184>] do_group_exit+0x44/0xc0 Mar 4 22:04:04 aleph kernel: [<fffffc0000335a9c>] get_signal_to_deliver+0x2fc/0x450 Mar 4 22:04:04 aleph kernel: [<fffffc00003167f4>] do_notify_resume+0xb4/0x570 Mar 4 22:04:04 aleph kernel: [<fffffc00003110cc>] work_pending+0x5c/0x70 Mar 4 22:04:04 aleph kernel: [<fffffc0000333f60>] __sigqueue_alloc+0x40/0xc0 Mar 4 22:04:04 aleph kernel: [<fffffc000038f7a0>] do_ioctl+0x30/0x90 Mar 4 22:04:04 aleph kernel: [<fffffc0000334954>] specific_send_sig_info+0xd4/0x110 Mar 4 22:04:04 aleph kernel: [<fffffc0000334a9c>] force_sig_info+0x8c/0xe0 Mar 4 22:04:04 aleph kernel: [<fffffc0000310bdc>] entMM+0x9c/0xc0 Mar 4 22:04:04 aleph kernel: Mar 4 22:04:04 aleph kernel: Code: a77df1b0 6b5b497f 27ba0039 23bdf664 c3ffffec 00000081 <00000806> 005efcfe Mar 4 22:04:04 aleph kernel: Fixing recursive fault but reboot is needed! Modules installed: ipv6 evdev snd_ice1724 snd_ice17xx_ak4xxx snd_ac97_codec ac97_bus snd_ak4114 snd_pt2258 snd_i2c snd_ak4xxx_adda snd_mpu401_uart snd_pcm_oss snd_pcm snd_page_alloc snd_mixer_oss snd_seq_dummy snd_seq_oss snd_seq_midi snd_rawmidi snd_seq_midi_event snd_seq snd_timer snd_seq_device snd soundcore pcspkr serio_raw tulip Sound devices: **** List of PLAYBACK Hardware Devices **** card 0: Revolution71 [M Audio Revolution-7.1], device 0: ICE1724 [ICE1724] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Revolution71 [M Audio Revolution-7.1], device 1: IEC1724 IEC958 [IEC1724 IEC958] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Revolution71 [M Audio Revolution-7.1], device 2: ICE1724 Surrounds [ICE1724 Surround PCM] Subdevices: 3/3 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 cat /proc/asound/cards: 0 [Revolution71 ]: ICE1724 - M Audio Revolution-7.1 M Audio Revolution-7.1 at 0x8480, irq 40 PCI devices: 00:03.0 Ethernet controller: Digital Equipment Corporation DECchip 21142/43 (rev 30) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr+ Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 255 (5000ns min, 10000ns max), Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 24 Region 0: I/O ports at 9000 [size=128] Region 1: Memory at 09259000 (32-bit, non-prefetchable) [size=128] Expansion ROM at 09200000 [disabled] [size=256K] Kernel driver in use: tulip Kernel modules: tulip 00:07.0 ISA bridge: Contaq Microsystems 82c693 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr+ Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 00:07.1 IDE interface: Contaq Microsystems 82c693 (prog-if 80 [Master]) Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr+ Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin A routed to IRQ 0 Region 0: I/O ports at 01f0 [size=8] Region 1: I/O ports at 03f4 [size=1] Region 2: [virtual] Memory at 00000170 (32-bit, non-prefetchable) [disabled] [size=8] Region 3: [virtual] Memory at 00000370 (type 3, non-prefetchable) [disabled] [size=1] Region 4: I/O ports at 9080 [size=16] Kernel driver in use: Cypress_IDE 00:07.2 IDE interface: Contaq Microsystems 82c693 (prog-if 00 []) Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr+ Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin B routed to IRQ 0 Region 0: I/O ports at 01f0 [size=8] Region 1: I/O ports at 03f4 [size=1] Region 2: [virtual] Memory at 00000170 (32-bit, non-prefetchable) [disabled] [size=8] Region 3: [virtual] Memory at 00000370 (type 3, non-prefetchable) [disabled] [size=1] Region 4: Memory at 09240000 (32-bit, non-prefetchable) [disabled] [size=64K] 00:07.3 USB Controller: Contaq Microsystems 82c693 (prog-if 10 [OHCI]) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr+ Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 248 Interrupt: pin A routed to IRQ 10 Region 0: Memory at 09258000 (32-bit, non-prefetchable) [size=4K] Kernel driver in use: ohci_hcd 00:0c.0 VGA compatible controller: Digital Equipment Corporation PBXGB [TGA2] (rev 22) (prog-if 00 [VGA controller]) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr+ Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR+ INTx- Latency: 255 (2000ns min, 16000ns max) Interrupt: pin A routed to IRQ 32 Region 0: Memory at 10000000 (32-bit, prefetchable) [size=128M] Expansion ROM at 09250000 [disabled] [size=32K] 00:14.0 PCI bridge: Digital Equipment Corporation DECchip 21152 (rev 02) (prog-if 00 [Normal decode]) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 255, Cache Line Size: 64 bytes Bus: primary=00, secondary=01, subordinate=01, sec-latency=255 I/O behind bridge: 00008000-00008fff Memory behind bridge: 09000000-090fffff Prefetchable memory behind bridge: 0000000009100000-00000000091fffff Secondary status: 66MHz- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- <SERR- <PERR- BridgeCtl: Parity- SERR- NoISA- VGA- MAbort- >Reset- FastB2B- PriDiscTmr- SecDiscTmr- DiscTmrStat- DiscTmrSERREn- 01:04.0 SCSI storage controller: QLogic Corp. ISP1020 Fast-wide SCSI (rev 05) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr+ Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 248, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 27 Region 0: I/O ports at 8000 [size=256] Region 1: Memory at 09000000 (32-bit, non-prefetchable) [size=4K] Expansion ROM at 09100000 [disabled] [size=64K] Kernel driver in use: qla1280 01:09.0 Multimedia audio controller: VIA Technologies Inc. VT1720/24 [Envy24PT/HT] PCI Multi-Channel Audio Controller (rev 01) Subsystem: VIA Technologies Inc. M-Audio Revolution 7.1 Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 248 Interrupt: pin A routed to IRQ 40 Region 0: I/O ports at 8480 [size=32] Region 1: I/O ports at 8400 [size=128] Capabilities: [80] Power Management version 1 Flags: PMEClk- DSI- D1- D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: ICE1724 Kernel modules: snd-ice1724 Cheerz Michael.

2 3

[alsa-devel] [Fwd: Re: Davinci ASoC support (patch)]
by lokowich 31 Mar '08

31 Mar '08

Vova Barinov wrote: > Mark Brown wrote: > >> On Fri, Mar 28, 2008 at 09:27:04AM -0500, lokowich wrote: >> >> >> >>> Forwarded from davinci-linux-open-source list: >>> >>> >> CCing in Vladimir Barinov who contributed the DaVinci support. >> >> >> >>> I've built this ASoC patch into my DaVinci git kernel (thanks for the >>> SENSORS tip), and can use aplay and speaker-test, but no sound. Figured >>> it may be a mixer/volume control issue, but the alsamixer is unable to >>> open for a failed ioctl on USBDEVFS_IOCTL. Not sure why there is a USB >>> dependency. Noted that there is no /sys/class/sound/mixer in my sysfs. >>> Any ideas? >>> >>> > How do you figure out that you can play sound? > Also what platform do you use - is it Davinci-EVM? what line output do > you use to hear the sound? > Please, post your dmesg log and cat /proc/interrupts too. > Excerpt from dmesg: Advanced Linux Sound Architecture Driver Version 1.0.16. ASoC version 0.13.2 AIC3X Audio Codec 0.1 asoc: aic3x <-> davinci-i2s mapping ok ALSA device list: #0: DaVinci EVM (aic3x) /proc/interrupts: CPU0 6: 0 AINTC dm644xresizer 8: 23608 AINTC davincifb 13: 6995 AINTC EMAC 22: 107 AINTC ide0 32: 39436 AINTC clockevent 33: 3 AINTC free-run counter 39: 217 AINTC i2c_davinci 40: 204 AINTC serial Err: 0 This is the DM6446 EVM. Note, my kernel has been patched with latest ALSA, to no avail. I hear the DAC pop during initialization on both the Line Out and Headphone jacks. >> It might be helpful to post dmesg output showing the sound support >> being loaded >> >> >>> Bernard Blackham wrote: >>> >>> >>>> Behme wrote: >>>> >>>> >>>> >>>>> While searching something different, I just found >>>>> >>>>> http://www.spinics.net/lists/alsa-devel/msg12535.html >>>>> >>>>> As I'm not subscribed to alsa-devel, I don't have the original post. >>>>> Maybe any of the authors can post it do this list as well? >>>>> >>>>> The patch mainly applies to our git. I fixed two small rejects in >>>>> Kconfig and Makefile as the patch is for newer kernel than our git. >>>>> Untested patch against our git in attachment. Anybody likes to test? >>>>> >>>>> >>>>> >>>> Well, I tried compiling it and it failed to build. It appears that it's >>>> built on a set of patches for Davinci posted to LAK, including this >>>> completely revamped EDMA support[1]: >>>> >>>> http://www.spinics.net/lists/arm-kernel/msg47450.html >>>> >>>> > Yep, it's needed. > >>>> So I had to change a few definitions, import a few functions from the new >>>> DMA support and managed to get it compiling and loading, but not yet working >>>> (tried various sample rates and formats): >>>> >>>> root@davinci-dvevm:/tst$ aplay test.wav >>>> Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo >>>> ALSA lib pcm_params.c:2152:(snd_pcm_hw_refine_slave) Slave PCM not usable >>>> aplay: set_params:878: Broken configuration for this PCM: no configurations >>>> available >>>> >>>> root@davinci-dvevm:/tst$ arecord -r 48000 -f S16_LE >>>> Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono >>>> ALSA lib pcm_params.c:2152:(snd_pcm_hw_refine_slave) Slave PCM not usable >>>> arecord: set_params:878: Broken configuration for this PCM: no >>>> configurations available >>>> >>>> >>>> And this is the extent of my knowledge of ALSA - I can continue digging, but >>>> maybe somebody else will beat me to it =) It's looking promising though. >>>> >>>> Attached is the patch I have so far that applies against current git. This >>>> is not for going upstream, but just for testing. Notes: >>>> - Apply patch >>>> - Enable SND_SOC, SND_DAVINCI_SOC, SND_DAVINCI_SOC_I2S, SND_DAVINCI_SOC_EVM >>>> and SND_SOC_TLV320AIC3X. >>>> - Make sure CONFIG_I2C_DAVINCI is enabled, but disable the >>>> CONFIG_SENSORS_TLV320* options. >>>> - Build. >>>> - Load the snd-soc-evm module. >>>> >>>> Bernard. >>>> >>>> [1] These patches appear to have gone absolutely nowhere beyond that list. I >>>> can't find them in any tree. But they are cleaner and more conforming to the >>>> Linux Kernel style than what's currently in git. >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> >>>> _______________________________________________ >>>> Davinci-linux-open-source mailing list >>>> Davinci-linux-open-source(a)linux.davincidsp.com >>>> http://linux.davincidsp.com/mailman/listinfo/davinci-linux-open-source >>>> >>>> >>>> >>> _______________________________________________ >>> 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 > > >

1 0

Re: [alsa-devel] Any wigets for ADC to DAC ?
by Mark Brown 31 Mar '08

31 Mar '08

On Mon, Mar 31, 2008 at 12:32:21PM +0200, Alexandre BOUIN wrote: > Will be there a support for a such feature in ASoC v2 ? Given that more and more codecs support this sort of thing it's likely that support for digital paths will get implemented at some point. No specific ETA for it, though. Note that if there is no specific reason to use a digital bypass path (such as filtering that is only supported on the digital path or a lack of a suitable analogue path) then it is normally more power efficient to use an analogue bypass path than a digital one.

1 0

Re: [alsa-devel] Any wigets for ADC to DAC ?
by Mark Brown 31 Mar '08

31 Mar '08

On Fri, Mar 28, 2008 at 04:39:48PM +0100, Alexandre BOUIN wrote: > In intern, we're still using WM8900 codec. We'd like to know in ASoC layer, > there is a widget to declare a possible audio path between ADC and DAC. If there were a direct connection the appropriate thing would just be to declare it (in that particular chip there is actually a digitial mixer in there). Using such a path is not particularly supported by ASoC v1 since it treats ADCs and DACs as sinks and sources only.

2 1

[alsa-devel] [PATCH]Should use HDA_OUTPUT instead of HDA_INPUT to mute pin 15 of ALC880
by Jiang zhe 31 Mar '08

31 Mar '08

Hi, To mute the output of Pin widget 15 in ALC880, we should use the HDA_OUTPUT. However, current code looks like : snd_hda_codec_amp_stereo(codec, 0x15, HDA_INPUT, 0, HDA_AMP_MUTE, bits); It may be a misspelling. Best regards! Signed-off-by: Jiang zhe <zhe.jiang(a)intel.com>

1 0

Re: [alsa-devel] [regression] 2.6.25-rc4 snd-es18xx broken on Alpha
by Rene Herman 31 Mar '08

31 Mar '08

On 10-03-08 08:34, Michael Cree wrote: > Bob Tracy wrote: >> Supposedly with the ES1888, dma1 is for capture, dma2 is for playback. >> dma2 == 5 is a 16-bit channel, yes? That could explain much... It is, but what would it explain? You're only having playback problems, right? >> >> As for the values "chosen" for dma1 and dma2, they are the ones that >> keep showing up in the Alpha sound "howto" postings/documents. > > Running 'show config' in SRM on my Alphas (PWS500au and XP1000) reveals > the settings (except possibly for mpu_port) for the ESS1888 sound chip. > I recall that they are as reported in Alpha sound howtos. Can it be forced to use dma2=0 (an 8-bit channel, and the usual capture channel on es18xx)? However, that might not be the issue anyway: > I hadn't been using the ESS1888 for awhile, but have just tried it out > since you reported problems. I am running kernel 2.6.24.3 and Debian > testing on the XP1000. I tried playing a number of wav files with > alsa's aplay, sox's play and with mocp. I found that aplay and mocp > worked reliably through the ESS1888. Sox's play on some files did play > back a small extra segment of the file - particularly on short files and > usually some section near the end of the file - once it had completed > playing the file once; maybe this is what you are also observing. This sounds very suspiciously like a difference with playing through the native ALSA interface and the OSS emulaion. Could you and/or Bob confirm that sox is using the OSS emulation and not ALSA natively? I could very well imagine the ALSA OSS emulation being broken on Alpha. I doubt any of teh developers has an Alpha. And if aplay works correctly this seems very likely. "sox" _can_ use ALSA natively as well by the way (see manpage for the version you have installed). > More onerously, my testing eventually ended in a complete system lock up! > I ran play (or was it aplay - sorry can't remember now) Vital if play is using the OSS interface... > and the system locked up. Got a response with ping across the network but > couldn't log in via ssh. Have been playing mplayer through the ESS1888 > for the last couple of days (don't like it though - the sound quality of > the ESS1888 is not good enough for my ears) and haven't had another one > of those lockups since. Mmm. Rene.

7 50

[alsa-devel] ATI HDMI with 780/sb700 chipset: no audio
by mikp 30 Mar '08

30 Mar '08

I am trying to get HDMI audio from a ECS 780G mainboard with sb700 and built-in graphics Chipset: "ATI Radeon HD 3200 Graphics" (ChipID = 0x9610). The HDMI video is working with the xf86-video-ati driver. aplay -l **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDA ATI HDMI], device 0: HDA Generic [HDA Generic] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI_1 [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 and from hwinfo: 31: PCI 14.2: 0403 Audio device [Created at pci.301] UDI: /org/freedesktop/Hal/devices/pci_1002_4383 Unique ID: 5Dex.2J+_taPgpb9 SysFS ID: /devices/pci0000:00/0000:00:14.2 SysFS BusID: 0000:00:14.2 Hardware Class: sound Model: "Elitegroup SBx00 Azalia" Vendor: pci 0x1002 "ATI Technologies Inc" Device: pci 0x4383 "SBx00 Azalia" SubVendor: pci 0x1019 "Elitegroup Computer Systems" SubDevice: pci 0x2816 Driver: "HDA Intel" Driver Modules: "snd_hda_intel" Memory Range: 0xfe7f4000-0xfe7f7fff (rw,non-prefetchable) IRQ: 16 (13305 events) Module Alias: "pci:v00001002d00004383sv00001019sd00002816bc04sc03i00" Driver Info #0: Driver Status: snd_hda_intel is active Driver Activation Cmd: "modprobe snd_hda_intel" Config Status: cfg=new, avail=yes, need=no, active=unknown cat /proc/asound/card1/codec\#0 Codec: ATI ATI RS690/780 HDMI Address: 0 Vendor Id: 0x1002791a Subsystem Id: 0x791a00 Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x40]: 48000 bits [0x2]: 16 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x201: Stereo Digital Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 Node 0x03 [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x0894: OUT Detect R/L Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x02 running alsamixer -c1 i can unmute the IEC958, but i get no HDMI audio. Note the "...Digital Out at Int HDMI" _not_ SPDIF Out at Int HDMI. In order to get the 'card1 with IEC958' to show up i had to add my pci-id to alsa-driver-hg20080323/sound/pci/hda/hda_intel.c /* ATI HDMI */ { PCI_DEVICE(0x1002, 0x4383), .driver_data = AZX_DRIVER_ATIHDMI }, The original hda_intel.c has { 0x1002, 0x4383, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATI }, Any help getting this working appreciated. Thanks.

1 0

[alsa-devel] Cannot change volume on Plantronics Audio 910 USB Headset
by cklein＠mpi-sb.mpg.de 29 Mar '08

29 Mar '08

Hi, I recently got a Plantronics .audio 910 USB headset. It is correctly detected by snd-usb-audio and works, but I cannot change the volume of the PCM output or the MIC input. When executing alsamixer -c 1, the volumes seem to be locked and cannot be changed as usual. Unfortunately the microphone volume is at 0, leading to very poor quality of the recorded voice (especially in Skype, which seems to try to compensate the low volume and fails miserably). Does anyone know how to fix this? I found similar reports on the internet for various usb headsets, but no real fix for the situation. I already asked on alsa-users some weeks ago, but did not get any response. I tried the headset on three different PCs, all with the same result. A desktop at work running Fedora 7, a Thinkpad R50p running debian/testing and the 2.6.22 kernel provided in debian/testing. The third one is a Dell precision M65 with debian/unstable. There, I used a vanilla 2.6.24.2 kernel and latest alsa drivers from the mercurial repository when I asked on alsa-users, but in the meantime I upgraded to 2.6.24.3 and hg snapshots from 2008-03-13 06:41 (alsa-driver) and 2008:03:17 05:23 (alsa-kernel), but still got the same behaviour. On all I get some message about "no or invalid class specific endpoint descriptor" in the kernel log (see below). When loading the snd-usb-audio module, dmesg shows the following kernel output (on the M65): usb 2-2: new full speed USB device using uhci_hcd and address 5 usb 2-2: configuration #1 chosen from 1 choice input: Plantronics Plantronics BT Adapter as /devices/pci0000:00/0000:00:1d.1/usb2/2-2/2-2:1.3/input/input10 input: USB HID v1.11 Device [Plantronics Plantronics BT Adapter] on usb-0000:00:1d.1-2 usbcore: registered new interface driver usbhid drivers/hid/usbhid/hid-core.c: v2.6:USB HID core driver ALSA /usr/src/kernel-modules/alsa-driver-d20080313_0641-k20080317_0523/usb/usbaudio.c:2787: 5:1:1 : no or invalid class specific endpoint descriptor ALSA /usr/src/kernel-modules/alsa-driver-d20080313_0641-k20080317_0523/usb/usbaudio.c:2787: 5:2:1 : no or invalid class specific endpoint descriptor usbcore: registered new interface driver snd-usb-audio I posted the output of alsa-info.sh and lsusb -v on pastebin (both taken on the M65). alsa-info.sh: http://pastebin.ca/915715 lsusb -v: http://pastebin.ca/915723 -- CU, Christian Klein http://www.garfield.com/comics/comics_archives_strip.html?1979-ga790520 Homepage: http://www.mpi-sb.mpg.de/~cklein

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[alsa-devel] [BUG] USB sounds card doesn't work - Device 0x4fa:0x4201
by Natalie Protasevich 28 Mar '08

28 Mar '08

Hello ALSA developers, This problem with the USB sounds card has been reported a while ago, and just been confirmed with latest kernel. ==================================================================== $ cat /proc/asound/cards 0 [I82801DBICH4 ]: ICH4 - Intel 82801DB-ICH4 Intel 82801DB-ICH4 with AD1981B at 0xc0000c00, irq 11 1 [U0x4fa0x4201 ]: USB-Audio - USB Device 0x4fa:0x4201 USB Device 0x4fa:0x4201 at usb-0000:00:1d.1-2, full speed If I attempt to play a song using mpg123 or xmms, I get nothing, only maybe click at beggining. I'm using in-kernel OSS emulation. pavel@amd:/mnt/mp3.ptacek/Tanita_Tikaram$ cat /proc/asound/card1/pcm0p/info card: 1 device: 0 subdevice: 0 stream: PLAYBACK id: USB Audio name: USB Audio subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 pavel@amd:/mnt/mp3.ptacek/Tanita_Tikaram$ I tried unmuting everything using aumix. cat /bin/bash > /dev/dsp1 fails to produce any sound, too. =================================================================== Full bug report is here : http://bugzilla.kernel.org/show_bug.cgi?id=6576 It sure needs some helping hand, maybe someone can take a look into this. Thanks, --Natalie

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[alsa-devel] USB sound card fails to resume from hibernation due to duplicate filename card1/WARNING: sysfs_add_ons, emits symbols
by Avuton Olrich 28 Mar '08

28 Mar '08

After suspending, the usbdisk sound card alsa driver breaks and does not play any further. I believe this is might be due to the trace below from the dmesg. This definitely begins to occur between v2.6.23 and v2.6.23-rc2. I couldn't bisect it down to a single commit. This fails with with all kernels up to at least today's git. Full links at the bottom. Linux tulip 2.6.25-rc6 #6 PREEMPT Wed Mar 19 22:16:23 PDT 2008 i686 AMD Athlon(tm) XP 2700+ AuthenticAMD GNU/Linux Gnu C 4.2.3 Gnu make 3.81 binutils 2.18 util-linux 2.13.1 mount 2.13.1 module-init-tools 3.4 e2fsprogs 1.40.8 xfsprogs 2.9.7 PPP 2.4.4 Linux C Library 2.7 Dynamic linker (ldd) 2.7 Procps 3.2.7 Net-tools 1.60 Kbd 1.13 Sh-utils 6.9 udev 119 Modules Loaded netconsole configfs snd_pcm_oss snd_mixer_oss Relevant looking parts of the dmesg (full dmesg link below): [ 3895.879517] usb 1-4: New USB device found, idVendor=2001, idProduct=f103 [ 3895.879517] usb 1-4: New USB device strings: Mfr=0, Product=0, SerialNumber=0 [ 3895.879517] usb 3-1: USB disconnect, address 2 [ 3896.252863] usb 3-1: new full speed USB device using ohci_hcd and address 3 [ 3896.452987] usb 3-1: configuration #1 chosen from 1 choice [ 3896.458877] ALSA /usr/src/linux-2.6/sound/usb/usbaudio.c:2798: 3:1:1: add audio endpoint 0x1 [ 3896.470877] ALSA /usr/src/linux-2.6/sound/usb/usbaudio.c:2798: 3:2:1: add audio endpoint 0x82 [ 3896.482878] ALSA /usr/src/linux-2.6/sound/usb/usbmixer.c:988: [13] FU [Mic Playback Switch] ch = 1, val = 0/1/1 [ 3896.521857] ALSA /usr/src/linux-2.6/sound/usb/usbmixer.c:405: cannot set ctl value: req = 0x4, wValue = 0x200, wIndex = 0xd00, type = 4, data = 0x18/0x0 [ 3896.554854] ALSA /usr/src/linux-2.6/sound/usb/usbmixer.c:988: [13] FU [Mic Playback Volume] ch = 1, val = 0/12240/384 [ 3896.554870] ALSA /usr/src/linux-2.6/sound/usb/usbmixer.c:988: [9] FU [Speaker Playback Switch] ch = 1, val = 0/1/1 [ 3896.593850] ALSA /usr/src/linux-2.6/sound/usb/usbmixer.c:405: cannot set ctl value: req = 0x4, wValue = 0x201, wIndex = 0x900, type = 4, data = 0x18/0x0 [ 3896.608842] ALSA /usr/src/linux-2.6/sound/usb/usbmixer.c:988: [9] FU [Speaker Playback Volume] ch = 2, val = -7264/-16/48 [ 3896.608859] ALSA /usr/src/linux-2.6/sound/usb/usbmixer.c:988: [10] FU [Mic Capture Switch] ch = 1, val = 0/1/1 [ 3896.647835] ALSA /usr/src/linux-2.6/sound/usb/usbmixer.c:405: cannot set ctl value: req = 0x4, wValue = 0x200, wIndex = 0xa00, type = 4, data = 0x18/0x0 [ 3896.680836] ALSA /usr/src/linux-2.6/sound/usb/usbmixer.c:988: [10] FU [Mic Capture Volume] ch = 1, val = 0/6096/384 [ 3896.680849] ALSA /usr/src/linux-2.6/sound/usb/usbmixer.c:988: [10] FU [Auto Gain Control] ch = 1, val = 0/1/1 [ 3896.680880] sysfs: duplicate filename 'card1' can not be created [ 3896.680891] ------------[ cut here ]------------ [ 3896.680893] WARNING: at /usr/src/linux-2.6/fs/sysfs/dir.c:424 sysfs_add_one+0x82/0xa0() [ 3896.680896] Modules linked in: netconsole configfs snd_pcm_oss snd_mixer_oss [ 3896.680903] Pid: 174, comm: khubd Not tainted 2.6.25-rc6 #6 [ 3896.680906] [<c011b0ef>] warn_on_slowpath+0x5f/0x90 [ 3896.680914] [<c0115fde>] __wake_up_common+0x3e/0x70 [ 3896.680918] [<c011b85b>] wake_up_klogd+0x3b/0x40 [ 3896.680922] [<c017d591>] find_inode+0x31/0x60 [ 3896.680927] [<c01a8c40>] sysfs_ilookup_test+0x0/0x10 [ 3896.680930] [<c011c0cb>] printk+0x1b/0x20 [ 3896.680933] [<c01a9012>] sysfs_add_one+0x82/0xa0 [ 3896.680937] [<c01a9dfc>] sysfs_create_link+0x8c/0x100 [ 3896.680940] [<c02f2cb4>] device_add+0x254/0x460 [ 3896.680946] [<c02f2f58>] device_create+0x88/0xc0 [ 3896.680949] [<c038aa06>] snd_card_register+0x276/0x380 [ 3896.680954] [<c03b2ae9>] usb_audio_probe+0x459/0xa00 [ 3896.680959] [<c01a8c40>] sysfs_ilookup_test+0x0/0x10 [ 3896.680963] [<c0342c66>] usb_probe_interface+0xb6/0x140 [ 3896.680969] [<c02f4ca0>] __device_attach+0x0/0x10 [ 3896.680973] [<c02f4ba1>] driver_probe_device+0x81/0x180 [ 3896.680976] [<c043ecc9>] klist_next+0x69/0xd0 [ 3896.680981] [<c02f4ca0>] __device_attach+0x0/0x10 [ 3896.680984] [<c02f3e76>] bus_for_each_drv+0x46/0x70 [ 3896.680988] [<c02f4d5b>] device_attach+0x7b/0x80 [ 3896.680991] [<c02f4ca0>] __device_attach+0x0/0x10 [ 3896.680994] [<c02f3e07>] bus_attach_device+0x47/0x70 [ 3896.680997] [<c02f2dfb>] device_add+0x39b/0x460 [ 3896.681000] [<c0340b4d>] usb_set_configuration+0x30d/0x5d0 [ 3896.681004] [<c02f4ca0>] __device_attach+0x0/0x10 [ 3896.681007] [<c0349186>] generic_probe+0x76/0xb0 [ 3896.681012] [<c02f4ca0>] __device_attach+0x0/0x10 [ 3896.681015] [<c0342a20>] usb_probe_device+0x30/0x40 [ 3896.681018] [<c02f4ba1>] driver_probe_device+0x81/0x180 [ 3896.681021] [<c043ecc9>] klist_next+0x69/0xd0 [ 3896.681024] [<c02f4ca0>] __device_attach+0x0/0x10 [ 3896.681027] [<c02f3e76>] bus_for_each_drv+0x46/0x70 [ 3896.681030] [<c02f4d5b>] device_attach+0x7b/0x80 [ 3896.681033] [<c02f4ca0>] __device_attach+0x0/0x10 [ 3896.681036] [<c02f3e07>] bus_attach_device+0x47/0x70 [ 3896.681039] [<c02f2dfb>] device_add+0x39b/0x460 [ 3896.681042] [<c033b07b>] usb_configure_device+0x5b/0x100 [ 3896.681045] [<c033b33b>] usb_new_device+0x5b/0x180 [ 3896.681048] [<c033c957>] hub_thread+0x637/0xd10 [ 3896.681051] [<c012ddb0>] autoremove_wake_function+0x0/0x40 [ 3896.681058] [<c033c320>] hub_thread+0x0/0xd10 [ 3896.681060] [<c012da82>] kthread+0x42/0x70 [ 3896.681063] [<c012da40>] kthread+0x0/0x70 [ 3896.681066] [<c0104b9f>] kernel_thread_helper+0x7/0x18 [ 3896.681071] ======================= [ 3896.681073] ---[ end trace 87ceb0cbe974168a ]--- [ 3896.689903] input: C-Media USB Headphone Set as /devices/pci0000:00/0000:00:02.1/usb3/3-1/3-1:1.3/input/input9 [ 3896.763158] input: USB HID v1.00 Device [C-Media USB Headphone Set ] on usb-0000:00:02.1-1 [ 3896.763218] usb 3-1: New USB device found, idVendor=0d8c, idProduct=000c [ 3896.763221] usb 3-1: New USB device strings: Mfr=0, Product=1, SerialNumber=0 [ 3896.763224] usb 3-1: Product: C-Media USB Headphone Set /proc/modules netconsole 8064 0 - Live 0xf88eb000 configfs 21396 2 netconsole, Live 0xf88f3000 snd_pcm_oss 45280 0 - Live 0xf88fa000 snd_mixer_oss 14784 1 snd_pcm_oss, Live 0xf88e6000 http://avuton.googlepages.com/dmesg-2.6.25-rc6-032008 http://avuton.googlepages.com/iomem-032008 http://avuton.googlepages.com/ioports-032008 http://avuton.googlepages.com/lspci-vvv-032008 http://avuton.googlepages.com/tulip-config -- avuton -- Anyone who quotes me in their sig is an idiot. -- Rusty Russell.

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[alsa-devel] asoc driver for ad1888 available?
by Tzachi Perelstein 28 Mar '08

28 Mar '08

Hello, I'm looking for an asoc ad1888 codec driver. Does anyone know if such driver available somewhere? I couldn't find one with google nor on linux-2.6-asoc.git. Can anyone suggest an alternative AC97 2.3 codec device with AC97- Link which has an available asoc driver? Thanks. - Tzachi

2 1

[alsa-devel] Accessing alsa user space buffer from external io type plugin
by sandeep lahane 27 Mar '08

27 Mar '08

Hello, I am debugging an userspace external io type plugin, written for an ARM based board where the plugin interacts with DSP directly using shared memory. Every time I have to send period_size data to the DSP, I have my own write/read methods to do so. I am having a problem accessing the alsa ring buffer, my transfer function looks like following. static char * temp; static snd_pcm_sframes_t alsa_transfer(snd_pcm_ioplug_t *io, const snd_pcm_channel_area_t *areas, snd_pcm_uframes_t offset, snd_pcm_uframes_t size) { ...... temp = malloc(period_size); memcpy(temp, unsigned char *)areas->addr + (areas->first + areas->step * offset) / 8, period_size); write_to_dsp(temp, period_size); ....... } All the book keeping is done properly, the first chunk of data is sent properly to DSP and pointer() returns correct value, next time when transfer is called, the memcpy gives segfault. ALSA buffer is an interleaved buffer and I am not using mmap mode. Any idea what might be wrong? When I do something like this: memcpy(temp, "test data test data", period_size); the data is sent properly to DSP, but alsa buffer data is not getting accessed properly. -- Regards, Sandeep.

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[alsa-devel] ESI Juli@/Ice1724
by Mark Trompell 27 Mar '08

27 Mar '08

Hi, when, trying to use audacity for recording, my system completly hangs. I wonder if this is the problem descriped in http://mailman.alsa-project.org/pipermail/alsa-devel/2007-April/000563.html http://mailman.alsa-project.org/pipermail/alsa-devel/2008-February/005987.h… and maybe https://bugtrack.alsa-project.org/alsa-bug/view.php?id=1860 Is it already fixed or is this bug still there? Regards Mark P.S I'm not subcribed to the list, so please cc me.

2 3

Re: [alsa-devel] ALSA driver for Native Instruments sound hardware
by kyr 27 Mar '08

27 Mar '08

O/H Daniel Mack ??????: > Hi, > > On Mar 21, 2008, at 2:07 PM, kyr wrote: >> I'd like to know if this driver support all of the 8 inputs/outputs >> channels, >> and if someone test it with xwax (xwax.co.uk). > > All inputs and outputs are supported, and the developer of xwax wanted > to test his software with this interface. However, I don't know about > his results or if he ever really did it. Maybe ask on their mailing list? > > Best regards, > Daniel > > Daniel hi, There is a xwax user with the audio8dj interface but he has some trouble to get it work well with xwax. The device is working almost well with plughw:1,0,0 for the first channel and plughw:1,0,1 for the second one. But with very high latency. Also refers that the alsamixer does not have any adjustments for in/outs. How can you adjust the input/output levels? and the frequency? 44100, 48000 or 96000Hz? Thanks in advance, Kyriakos

2 1

[alsa-devel] Getting wrong offset and size value alsa library
by Leonidas . 27 Mar '08

27 Mar '08

Hi, I am trying to make my user space audio driver work, in my drivers transfer function, I get wrong values of offset and size and because of this the alsa buffer is not accessible to me and I get a segfault in transfer. The offset should range from 0 to n-1 where n is the buffer size, is this correct? I can transfer the first chunk of data and first time the offset as well as size are correct, but when the transfer function is called again for second period, offset values and size values are wrong, they are some astronomical numbers. -leo

1 0

[alsa-devel] Vostro 1400 patch
by Martin van Es 26 Mar '08

26 Mar '08

Hi, I recently bought a Dell Vostro 1400 and thought I'd let you know my experiences with Alsa: I use Kubuntu Gutsy distribution as a base, but follow kernel at a close pace so I use compiled 2.6.24. This didn't support the hda-intel as I expected (don't remember exactly, but master/front/pcm/surround were all in the game but not as expected and I couldn't get built-in mic to work, but this may be due to too little testing). So I decided to google around and found the model=3stack, dell-3stack options and tried those. Still no luck. Next I dl'ed alsa 1.0.16 and noticed a mail describing a vostro 1400 PCI quirck patch (model dell-bios). So I tried that, but no luck. After that I experimented with all 3stack variants I could find in the code and concluded that dell-3stack is the best. I now have sound, master/front/pcm volume sliders that work but above all: built-in mic (Digital 1) recording (and skype) works! I don't know if it's of any value: I immediately flashed the standard A06 Bios to A07 when I found the update on the Dell site and can not tell if the dell-bios model worked for the preloaded A06 version bios. Hope this helps. If there's anything else I can provide I'd be more than happy to collect the data. Regards, Martin -- if but was of any use, it would be a logic operator

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Re: [alsa-devel] PXA2xx/3xx AC97 Driver(s)
by Mark Brown 26 Mar '08

26 Mar '08

On Wed, Mar 26, 2008 at 01:52:39PM +0000, Oliver Ford wrote: > The machine no longer reboots out of linux though. I assume there were > some changes in 2.6.25-rc4 mainline that aren't in the asoc-v2-dev yet. > I will wait for the pxa3xx stuff to be merged into the mainline for > this. What version of the mainline kernel is the latest asoc-v2-dev > branch based on. (I'm still a little hazy on the git stuff I'm afraid). The last version that was merged in appears to be 2.6.25-rc2 (say 'git log' then search for 'Merge.*v2\.6'). For production use I would recommend going with ASoC v1 at the moment. The PXA3xx support should be straightforward to backport (I wouldn't be suprised if the patches introducing it could just applied against mainline but I've not actually tried yet). > > I know that there are a number of users currently using the tree for > > that use case so the asoc-v2-dev branch should hopefully work for you. > > If you run into trouble please let me know. > The WM97xx touchscreen driver only appears to be in the linux-2.6-touch > repository (dev branch, iirc), rather than linux-2.6-asoc where the > pxa3xx support is. Since I want both the wm97xx touchscreen and pxa3xx > support, is there a simple way I can get the necessary changes in one > applied to the other? You can track both multiple git trees using the remotes feature of git - see git-remote(1) for usage information. For example, this: git remote add wolfson-touch \ git://opensource.wolfsonmicro.com/linux-2.6-touch will tell git to track the Wolfson touch drivers using the name 'wolfson-touch'. If you do that and then say: git remote update git merge wolfson-touch/dev it will fetch the current state of all remote trees it knows about and then merge the current dev branch of the Wolfson touch drivers git into your current working copy. The touchscreen drivers and git usage are both rather off-topic for either list, though - probably best to follow up off-list or on linux-input.

1 0

[alsa-devel] Plugin default and sw_params
by Emmanuel Milou 26 Mar '08

26 Mar '08

Hi all, I have some questions about the /default/ plugin. I am developping a softphone based on ALSA on Ubuntu 7.10. I use read/write IO. When I open a device with the /plughw /plugin, quite everything works fine, the quality sound is good. But with the /default/ plugin or the /dmix/ plugin ( the user can choose one of them ), I can't set my own hw parameters or sw parameters. Actually I set my own value of period size, buffer size, start threshold, stop threshold, but when I check in /proc/asound/cardx/pcm0p/sub0/sw_params or /proc/asound/cardx/pcm0p/sub0/hw_params, the values are totally different. For instance, the start threshold is set to 1 even if I set it to period_size. As a result, I get crackling sound. I suspect this is due to buffer underrun. Maybe if I could change the start threshold or the period size, the sound would be better. So my question: is it possible to modify this values for /default/ plugin without changing the alsa config file ( asoundrc ) ? Thank you very much Manu

1 0

[alsa-devel] snd-bt87x
by Bizon 26 Mar '08

26 Mar '08

Hello! I have WinTV TV card based on BT878A, correctly detected by driver. But there is no sound. I have just something like a single tone. How to fix it? For compare i have also Flyvideo 98 card based on BT878, detected as unknown - it works well. Here is it's lsmod : 00:0b.1 0480: 109e:0878 (rev 02) Subsystem: 1852:1852 Thanks Karol

1 0

[alsa-devel] [PATCH] hda: Add 5.1 support for second headphone jack
by Matthew Ranostay 25 Mar '08

25 Mar '08

Several 92hd7xxx and STAC9228 laptops have multiple headphone jacks, the second headphone jack should be used for the 5.1 surround sound. Add support for 'Headphone as Line Out' switch, which allows it be used in 5.1 surround sound. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 547e051a88b5 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Wed Mar 12 13:12:15 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Tue Mar 18 10:45:08 2008 -0400 @@ -179,6 +179,7 @@ struct sigmatel_spec { /* i/o switches */ unsigned int io_switch[2]; unsigned int clfe_swap; + unsigned int hp_switch; unsigned int aloopback; struct hda_pcm pcm_rec[2]; /* PCM information */ @@ -1251,7 +1252,7 @@ static unsigned int ref92hd73xx_pin_conf static unsigned int dell_m6_pin_configs[13] = { 0x0321101f, 0x4f00000f, 0x4f0000f0, 0x90170110, - 0x03a11020, 0x03011050, 0x4f0000f0, 0x4f0000f0, + 0x03a11020, 0x0321101f, 0x4f0000f0, 0x4f0000f0, 0x4f0000f0, 0x90a60160, 0x4f0000f0, 0x4f0000f0, 0x4f0000f0, }; @@ -2029,6 +2030,33 @@ static void stac92xx_auto_set_pinctl(str AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type); } +#define stac92xx_hp_switch_info snd_ctl_boolean_mono_info + +static int stac92xx_hp_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + + ucontrol->value.integer.value[0] = spec->hp_switch; + return 0; +} + +static int stac92xx_hp_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + + spec->hp_switch = ucontrol->value.integer.value[0]; + + /* check to be sure that the ports are upto date with + * switch changes */ + codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); + + return 1; +} + #define stac92xx_io_switch_info snd_ctl_boolean_mono_info static int stac92xx_io_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -2099,6 +2127,15 @@ static int stac92xx_clfe_switch_put(stru return 1; } + +#define STAC_CODEC_HP_SWITCH(xname) \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = 0, \ + .info = stac92xx_hp_switch_info, \ + .get = stac92xx_hp_switch_get, \ + .put = stac92xx_hp_switch_put, \ + } #define STAC_CODEC_IO_SWITCH(xname, xpval) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ @@ -2124,6 +2161,7 @@ enum { STAC_CTL_WIDGET_VOL, STAC_CTL_WIDGET_MUTE, STAC_CTL_WIDGET_MONO_MUX, + STAC_CTL_WIDGET_HP_SWITCH, STAC_CTL_WIDGET_IO_SWITCH, STAC_CTL_WIDGET_CLFE_SWITCH }; @@ -2132,6 +2170,7 @@ static struct snd_kcontrol_new stac92xx_ HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), STAC_MONO_MUX, + STAC_CODEC_HP_SWITCH(NULL), STAC_CODEC_IO_SWITCH(NULL, 0), STAC_CODEC_CLFE_SWITCH(NULL, 0), }; @@ -2396,6 +2435,14 @@ static int stac92xx_auto_create_multi_ou } } + if (cfg->hp_outs > 1) { + err = stac92xx_add_control(spec, + STAC_CTL_WIDGET_HP_SWITCH, + "Headphone as Line Out Switch", 0); + if (err < 0) + return err; + } + if (spec->line_switch) { nid = cfg->input_pins[AUTO_PIN_LINE]; pincap = snd_hda_param_read(codec, nid, @@ -3144,6 +3191,7 @@ static void stac92xx_hp_detect(struct hd { struct sigmatel_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; + int nid = cfg->hp_pins[cfg->hp_outs - 1]; int i, presence; presence = 0; @@ -3154,11 +3202,15 @@ static void stac92xx_hp_detect(struct hd for (i = 0; i < cfg->hp_outs; i++) { if (presence) break; + if (spec->hp_switch && cfg->hp_pins[i] == nid) + break; presence = get_hp_pin_presence(codec, cfg->hp_pins[i]); } if (presence) { /* disable lineouts, enable hp */ + if (spec->hp_switch) + stac92xx_reset_pinctl(codec, nid, AC_PINCTL_OUT_EN); for (i = 0; i < cfg->line_outs; i++) stac92xx_reset_pinctl(codec, cfg->line_out_pins[i], AC_PINCTL_OUT_EN); @@ -3167,6 +3219,8 @@ static void stac92xx_hp_detect(struct hd AC_PINCTL_OUT_EN); } else { /* enable lineouts, disable hp */ + if (spec->hp_switch) + stac92xx_set_pinctl(codec, nid, AC_PINCTL_OUT_EN); for (i = 0; i < cfg->line_outs; i++) stac92xx_set_pinctl(codec, cfg->line_out_pins[i], AC_PINCTL_OUT_EN); @@ -3174,6 +3228,8 @@ static void stac92xx_hp_detect(struct hd stac92xx_set_pinctl(codec, cfg->speaker_pins[i], AC_PINCTL_OUT_EN); } + if (!spec->hp_switch && cfg->hp_outs > 1 && presence) + stac92xx_set_pinctl(codec, nid, AC_PINCTL_OUT_EN); } static void stac92xx_pin_sense(struct hda_codec *codec, int idx) @@ -3427,6 +3483,7 @@ again: switch (spec->multiout.num_dacs) { case 0x3: /* 6 Channel */ + spec->multiout.hp_nid = 0x17; spec->mixer = stac92hd73xx_6ch_mixer; spec->init = stac92hd73xx_6ch_core_init; break;

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Re: [alsa-devel] PATCH - ESI Juli driver
by Demian Martin 25 Mar '08

25 Mar '08

Pavel: I want to thank you for getting the Juli@ driver sorted out. Its all working right and I can play the 176.4 KHz 24 bit files now properly. Along with 96K, 88.2K 48K and 44.1K. I had to borrow an Apogee 192K mini-dac to test the system. The inexpensive DAC I got off eBay from China was never tested properly for high bit rate files and let me off track until I got a known DAC to test with. The unique clocking scheme of the Juli@ card should give lower jitter and better audio than the other solutions since the clock isn’t generated from a PLL. It’s a relative bargain compared to a Hammerfall or a Lynx card. And now with a good Linux driver very useable. -Demian Demian Martin Product Design Service 784 Cary Drive San Leandro, CA 94577 209 613 6990 No virus found in this outgoing message. Checked by AVG. Version: 7.5.519 / Virus Database: 269.21.8/1340 - Release Date: 3/23/2008 6:50 PM

1 0

[alsa-devel] Wrong offset, step and size passed from alsa-lib layers to external io type plugin
by Pharaoh . 25 Mar '08

25 Mar '08

Hi, In my external io type plugin, the alsa buffer is not accessible in the transfer function, symptom of this is I am unable to memcpy period size data from alsa buffer to my local buffers. after debugging a little I found out that values of offset, size and areas->step are wrong. #aplay --period-size =1024 --buffer-size=8192 test.wav static snd_pcm_sframes_t alsa_transfer(snd_pcm_ioplug_t *io, const snd_pcm_channel_area_t *areas, snd_pcm_uframes_t offset, snd_pcm_uframes_t size) I get following, Buffer size = 32768 bytes Period size = 4096 bytes *ALSA buffer base addr = 0x2c6e0 //wrong Offset = 0xbeb4c7dc **//wrong* *size = 1063777280 **//wrong* step = 32 What could be the reason? -Pharaoh.

1 0

[alsa-devel] ALSA Debugging
by Vinod Koul 24 Mar '08

24 Mar '08

Hello, I am new to ALSA and Linux. I was trying to understand the ALSA architecture. I have Linux Box with Intel sound card. with Suse 10.1 on it. I launched the audio player (Real player) and audio plays fine. Then, I built the kernel and installed it. I put some printk's in intel8x0.cand control.c. (could see the my code was complied) Even though I changed volume muted etc couldn't see any of my printk in dmesg. One default printk in intel8x0.c was there, so I have put my print before that line as well, but couldn't see anything. Apologies If my problem is very basic. I read on forums that best way to learn is to put printk's. Can anyone help me with where I am doing incorrectly, Am I supposed to do it this way or some other way? Thanks in advance -Vinod

2 2

[alsa-devel] ALSA driver for Native Instruments sound hardware
by Daniel Mack 23 Mar '08

23 Mar '08

Hi, some months after the last posting, we finally managed to finish a new version of the ALSA driver for Native Instruments' external USB sound cards such as "Rig Kontrol 2", "Kore Kontroller", "Audio Kontrol 1" and "Audio 8 DJ". Compared to the last release, the following things were changed: - the streaming logic was completely reworked so that multiple substreams are handled correctly - a Linux input device layer was added to support the knobs, buttons and pedals on "Rig Kontrol 2" and "AudioKontrol 1" - workqueues were entirely removed from the code - most data structures are now allocated within one big struct to avoid multiple, smaller allocations - many more cleanups according to Clemens' reply on my last posting[1] The driver was properly tested with all the mentioned hardware on both record and playback channels. The patchset should apply to the current mecurial alsa-kernel tree. Please let me know what you think. Best regards, Daniel [1] http://article.gmane.org/gmane.linux.alsa.devel/34324/ Signed-off-by: Daniel Mack <daniel(a)caiaq.org> Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de>

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[alsa-devel] Buffer size/periods (Layla3G)
by Mark Hills 23 Mar '08

23 Mar '08

Hello, I am investigating problems I'm having with recording over long periods. I'm using a Layla3G and tested various ALSA versions including 1.0.14 and 1.0.16 on different machines. I've noticed that when I open the first pair of subdevices, my request for 1000ms buffer size is handled differently to other devices: $ ./test_buffers "hw:Layla3G,0,0" 1000 buffer_size=16384 period_size=128 $ ./test_buffers "hw:Layla3G,0,2" 1000 buffer_size=32768 period_size=256 $ ./test_buffers "hw:Layla3G,0,4" 1000 buffer_size=32768 period_size=256 $ ./test_buffers "hw:Layla3G,0,6" 1000 buffer_size=32768 period_size=256 If I try other buffer sizes, I can see is a limit to buffer size of 16384 samples on the ,0 subdevice, which is also affecting the chosen period size. But only on the first subdevice. Is there a reason, or is this a bug? Also, there seem to be failures at certain buffer sizes: $ ./test_buffers "hw:Layla3G,0,2" 429 buffer_size=18912 period_size=96 $ ./test_buffers "hw:Layla3G,0,2" 430 set_buffer_time_near: Invalid argument $ ./test_buffers "hw:Layla3G,0,2" 431 buffer_size=19008 period_size=96 Can anyone explain to me what's going on here, or is this also a bug? The overall problem I'm investigating is snd_pcm_readi() returning incomplete buffers after running for long periods until another snd_pcm_prepare(). It seems possible that this is related. Any help is greatly appreciated. Thanks, Mark

2 4

[alsa-devel] [PATCH] Support mic automute for Clevo M720R/SR
by Herton Ronaldo Krzesinski 22 Mar '08

22 Mar '08

Add support for mic automute in clevo-m720r ALC883 model, and rename it to more generic clevo-m720. Also change model entry in ALSA-Configuration.txt accordingly. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com> diff -p -up alsa-kernel/Documentation/ALSA-Configuration.txt.orig alsa-kernel/Documentation/ALSA-Configuration.txt --- alsa-kernel/Documentation/ALSA-Configuration.txt.orig 2008-03-20 16:51:03.000000000 -0300 +++ alsa-kernel/Documentation/ALSA-Configuration.txt 2008-03-20 16:52:00.000000000 -0300 @@ -886,7 +886,7 @@ Prior to version 0.9.0rc4 options had a 3stack-hp HP machines with 3stack (Lucknow, Samba boards) 6stack-dell Dell machines with 6stack (Inspiron 530) mitac Mitac 8252D - clevo-m720r Clevo laptop M720R + clevo-m720 Clevo M720 laptop series fujitsu-pi2515 Fujitsu AMILO Pi2515 auto auto-config reading BIOS (default) diff -p -up alsa-kernel/pci/hda/patch_realtek.c.orig alsa-kernel/pci/hda/patch_realtek.c --- alsa-kernel/pci/hda/patch_realtek.c.orig 2008-03-20 16:43:55.000000000 -0300 +++ alsa-kernel/pci/hda/patch_realtek.c 2008-03-20 16:49:22.000000000 -0300 @@ -201,7 +201,7 @@ enum { ALC888_3ST_HP, ALC888_6ST_DELL, ALC883_MITAC, - ALC883_CLEVO_M720R, + ALC883_CLEVO_M720, ALC883_FUJITSU_PI2515, ALC883_AUTO, ALC883_MODEL_LAST, @@ -6661,7 +6661,7 @@ static struct snd_kcontrol_new alc883_mi { } /* end */ }; -static struct snd_kcontrol_new alc883_clevo_m720r_mixer[] = { +static struct snd_kcontrol_new alc883_clevo_m720_mixer[] = { HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), @@ -7130,7 +7130,7 @@ static struct hda_verb alc883_mitac_verb { } /* end */ }; -static struct hda_verb alc883_clevo_m720r_verbs[] = { +static struct hda_verb alc883_clevo_m720_verbs[] = { /* HP */ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, @@ -7140,6 +7140,7 @@ static struct hda_verb alc883_clevo_m720 /* enable unsolicited event */ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN}, { } /* end */ }; @@ -7330,7 +7331,7 @@ static void alc883_tagra_unsol_event(str } /* toggle speaker-output according to the hp-jack state */ -static void alc883_clevo_m720r_automute(struct hda_codec *codec) +static void alc883_clevo_m720_hp_automute(struct hda_codec *codec) { unsigned int present; unsigned char bits; @@ -7342,11 +7343,33 @@ static void alc883_clevo_m720r_automute( HDA_AMP_MUTE, bits); } -static void alc883_clevo_m720r_unsol_event(struct hda_codec *codec, +static void alc883_clevo_m720_mic_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x18, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); +} + +static void alc883_clevo_m720_automute(struct hda_codec *codec) +{ + alc883_clevo_m720_hp_automute(codec); + alc883_clevo_m720_mic_automute(codec); +} + +static void alc883_clevo_m720_unsol_event(struct hda_codec *codec, unsigned int res) { - if ((res >> 26) == ALC880_HP_EVENT) - alc883_clevo_m720r_automute(codec); + switch (res >> 26) { + case ALC880_HP_EVENT: + alc883_clevo_m720_hp_automute(codec); + break; + case ALC880_MIC_EVENT: + alc883_clevo_m720_mic_automute(codec); + break; + } } /* toggle speaker-output according to the hp-jack state */ @@ -7605,7 +7628,7 @@ static const char *alc883_models[ALC883_ [ALC888_3ST_HP] = "3stack-hp", [ALC888_6ST_DELL] = "6stack-dell", [ALC883_MITAC] = "mitac", - [ALC883_CLEVO_M720R] = "clevo-m720r", + [ALC883_CLEVO_M720] = "clevo-m720", [ALC883_FUJITSU_PI2515] = "fujitsu-pi2515", [ALC883_AUTO] = "auto", }; @@ -7649,8 +7672,8 @@ static struct snd_pci_quirk alc883_cfg_t SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG), - SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720R), - SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720R), + SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720), + SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720), SND_PCI_QUIRK(0x1558, 0, "Clevo laptop", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), @@ -7794,17 +7817,17 @@ static struct alc_config_preset alc883_p .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, }, - [ALC883_CLEVO_M720R] = { - .mixers = { alc883_clevo_m720r_mixer }, - .init_verbs = { alc883_init_verbs, alc883_clevo_m720r_verbs }, + [ALC883_CLEVO_M720] = { + .mixers = { alc883_clevo_m720_mixer }, + .init_verbs = { alc883_init_verbs, alc883_clevo_m720_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, - .unsol_event = alc883_clevo_m720r_unsol_event, - .init_hook = alc883_clevo_m720r_automute, + .unsol_event = alc883_clevo_m720_unsol_event, + .init_hook = alc883_clevo_m720_automute, }, [ALC883_LENOVO_101E_2ch] = { .mixers = { alc883_lenovo_101e_2ch_mixer}, -- []'s Herton

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[alsa-devel] Kernel Oops unloading snd_ymfpci, 2.6.24
by Gerard Neil 22 Mar '08

22 Mar '08

Hi, I'm just upgraded to kernel 2.6.24 and I'm seeing a kernel OOPS every time I unload the snd_ymfpci module... Below is what I see in my log. My naive interpretation of this is that the system is barfing in the interrupt handler at the time it is trying to free the interrupt.... so maybe the interrupt hasn't been disabled properly before the free_irq() call? As I say, it's a naive interpretation. Anyway, I'd appreciate it if somebody could look this over and tell me what my next step should be. I'm available to debug if necessary. Thanks, Gerard Mar 18 00:16:58 {kern,info} ACPI: PCI Interrupt 0000:00:08.0[A] -> GSI 18 (level, low) -> IRQ 16 Mar 18 00:17:00 {kern,err} codec_ready: codec 0 is not ready [0x802a] Mar 18 00:17:00 {kern,alert} BUG: unable to handle kernel paging request at virtual address f8e10100 Mar 18 00:17:00 {kern,alert} printing eip: f8e2b430 *pde = 37770067 *pte = 00000000 Mar 18 00:17:00 {kern,emerg} Oops: 0000 [#1] SMP Mar 18 00:17:00 {kern,warning} Modules linked in: snd_ymfpci snd_seq snd_ac97_codec snd_pcm snd_opl3_lib snd_timer snd_hwdep snd_page_alloc snd_mpu401_uart snd_rawmidi snd_seq_device snd w83627hf hwmon_vid hwmon eeprom nfsd lockd nfs_acl sunrpc exportfs af_packet loop usbhid floppy ac97_bus soundcore ohci_hcd via_agp agpgart ehci_hcd skge uhci_hcd i2c_viapro i2c_core e100 Mar 18 00:17:00 {kern,warning} Mar 18 00:17:00 {kern,warning} Pid: 8543, comm: rmmod Not tainted (2.6.24-gentoo-r3 #1) Mar 18 00:17:00 {kern,warning} EIP: 0060:[<f8e2b430>] EFLAGS: 00210092 CPU: 1 Mar 18 00:17:00 {kern,warning} EIP is at snd_ymfpci_interrupt+0x10/0x1e0 [snd_ymfpci] Mar 18 00:17:00 {kern,warning} EAX: 00000010 EBX: f77f0200 ECX: e5a36980 EDX: f8e10000 Mar 18 00:17:00 {kern,warning} ESI: 00000010 EDI: dd74f000 EBP: c041e1d0 ESP: dd7a7e4c Mar 18 00:17:00 {kern,warning} DS: 007b ES: 007b FS: 00d8 GS: 0033 SS: 0068 Mar 18 00:17:00 {kern,emerg} Process rmmod (pid: 8543, ti=dd7a6000 task=e5a8aa60 task.ti=dd7a6000) Mar 18 00:17:00 {kern,emerg} Stack: f7c82960 00200286 000007b0 00000010 00000003 00200246 f77f0200 00000010 Mar 18 00:17:00 {kern,emerg} 00200282 c041e1d0 c01415b2 dd7b8000 1d7b8000 f7c57c4c e9e2a000 dd74f000 Mar 18 00:17:00 {kern,emerg} 00200216 dd74f000 e9e2a140 00002fff e9e2a000 f8e28d4f 00001fff f77f0d40 Mar 18 00:17:00 {kern,emerg} Call Trace: Mar 18 00:17:00 {kern,emerg} [<c01415b2>] free_irq+0x102/0x160 Mar 18 00:17:00 {kern,emerg} [<f8e28d4f>] snd_ymfpci_free+0x10f/0x150 [snd_ymfpci] Mar 18 00:17:00 {kern,emerg} [<f88bd6f3>] snd_device_free+0x63/0xb0 [snd] Mar 18 00:17:00 {kern,emerg} [<f88bd798>] snd_device_free_all+0x58/0x60 [snd] Mar 18 00:17:00 {kern,emerg} [<f88b96cd>] snd_card_do_free+0x2d/0xe0 [snd] Mar 18 00:17:00 {kern,emerg} [<f88ba361>] snd_card_free+0x91/0xa0 [snd] Mar 18 00:17:00 {kern,emerg} [<c022318b>] kref_put+0x2b/0x80 Mar 18 00:17:00 {kern,emerg} [<c03417cc>] wait_for_common+0x1c/0x140 Mar 18 00:17:00 {kern,emerg} [<f8e2801e>] snd_card_ymfpci_remove+0xe/0x20 [snd_ymfpci] Mar 18 00:17:00 {kern,emerg} [<c02309f6>] pci_device_remove+0x16/0x40 Mar 18 00:17:00 {kern,emerg} [<c027c6d4>] __device_release_driver+0x64/0xa0 Mar 18 00:17:00 {kern,emerg} [<c027cc21>] driver_detach+0xc1/0xd0 Mar 18 00:17:00 {kern,emerg} [<c027c235>] bus_remove_driver+0x75/0xa0 Mar 18 00:17:00 {kern,emerg} [<c0230ace>] pci_unregister_driver+0xe/0x70 Mar 18 00:17:00 {kern,emerg} [<c013f27a>] sys_delete_module+0x11a/0x1c0 Mar 18 00:17:00 {kern,emerg} [<c0114297>] do_page_fault+0xe7/0x620 Mar 18 00:17:00 {kern,emerg} [<c01553b7>] do_munmap+0x197/0x1f0 Mar 18 00:17:00 {kern,emerg} [<c0102b8e>] sysenter_past_esp+0x5f/0x85 Mar 18 00:17:00 {kern,emerg} [<c0340000>] unix_bind+0x170/0x2b0 Mar 18 00:17:00 {kern,emerg} ======================= Mar 18 00:17:00 {kern,emerg} Code: a8 c8 00 00 00 8b 47 10 8b 14 24 89 90 c0 00 00 00 8b 43 2c e9 40 ff ff ff 66 90 55 57 89 d7 56 53 83 ec 18 89 44 24 0c 8b 52 10 <8b> 82 00 01 00 00 85 c0 78 50 0f b7 42 04 0f b7 d8 f6 c3 01 74 Mar 18 00:17:00 {kern,emerg} EIP: [<f8e2b430>] snd_ymfpci_interrupt+0x10/0x1e0 [snd_ymfpci] SS:ESP 0068:dd7a7e4c Mar 18 00:17:00 {kern,warning} ---[ end trace 8697168af0baea4f ]---

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Re: [alsa-devel] PATCH - ESI Juli driver
by Demian Martin 21 Mar '08

21 Mar '08

Pavel; I have been following this closely since I have the juli@ card and have been having problems getting it to work at rates and bit depths other than 44.1/16 and 48/16 on the spdif output. I will be happy to provide a Juli@ card to support the work and keep this going. I’m working with a record company that is about to start releasing audio at 176.4/24 and the current playback options are very limited and expensive for that format. The Juli@ can open a lot of new options but it needs proper support. Let me know if you need it. Demian Martin Product Design Service No virus found in this outgoing message. Checked by AVG. Version: 7.5.519 / Virus Database: 269.21.7/1331 - Release Date: 3/16/2008 10:34 AM

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[alsa-devel] Accessing pcm samples from snd_pcm_channel_area_t in an io type plugin
by Pharaoh . 21 Mar '08

21 Mar '08

Hi, I am writing an external io type plugin for an ARM based board. In the transfer function of the plugin, I try to grab the audio samples in a temp buffer and send it to the DSP in chunks of period sizes. Somehow the the alsa buffer's audio samples are not copied to the temp buffer which I want to send further. What I do to copy the buffers, static snd_pcm_sframes_t my_transfer(snd_pcm_ioplug_t *io, const snd_pcm_channel_area_t *areas, snd_pcm_uframes_t offset, snd_pcm_uframes_t size) { char *temp = NULL; temp = malloc(io->period_size * frame_size); //For malloc, //I tried all the three approaches here one by one, including the malloc/memcpy one 1. memcpy(temp,areas->addr+ offset*io->channels*2, io->period_size * frame_size); 2. memcpy(temp, (char *)areas->addr + (areas->first + areas->step * offset) /8, io->period_size * frame_size); //This should work straight off.... 3. temp = (char *)areas->addr + (areas->first + areas->step * offset) / 8; //write temp to DSP here ......... } Somehow the empty buffers are being sent to DSP, when I try to send some data from a static array then it goes to DSP and is played well. Any idea what I am missing? -phraoh.

1 0

[alsa-devel] Buffer management in ALSA
by Aggarwal, Anuj 21 Mar '08

21 Mar '08

Hi All, Can anyone throw some light on the buffer management part in the ALSA framework? How and where it is implemented? And what is expected from the driver, which is hooking its code to the framework? Thanks in advance, Anuj Aggarwal

2 1

[alsa-devel] PATCH - ESI Juli driver
by Pavel Hofman 20 Mar '08

20 Mar '08

Hi Takashi, Juli support required some changes to ice1724.c/ice1712.h. Since Juli uses different clock scheme than most other ice1712/24 cards, I had to move all clock-specific code to card-specific routines. Most of them are redefined in juli.c. The driver is tested, only analog-in and digital-in monitors are not working. I have coded according to manufacturer's documentation, the same procedure is in OSS Juli driver. I described the issue in detail in the code, perhaps someone will do more research. The card itself has no HW master volume control. It would have to be implemented as a virtual control manipulating internal values of PCM and monitor volume controls. I will take a look at it later. For now, perhaps we could rename the PCM Volume to Master Volume, as most people will not need the monitoring anyway. I will leave it to your decision. Signed-off-by: Pavel Hofman <dustin(a)seznam.cz> * moving all clock-specific code to card-specific routines * support for ESI Juli * to-be-researched - monitoring of analog/digital inputs * missing - master volume control, will have to be implemented as virtual, no HW support. Thanks, Pavel.

2 18

[alsa-devel] [PATCH] Map clevo-m720r ALC883 model for Clevo M720SR
by Herton Ronaldo Krzesinski 20 Mar '08

20 Mar '08

Map clevo-m720r ALC883 model for Clevo M720SR. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com> diff -p -up alsa-kernel/pci/hda/patch_realtek.c.orig alsa-kernel/pci/hda/patch_realtek.c --- alsa-kernel/pci/hda/patch_realtek.c.orig 2008-03-20 16:32:34.000000000 -0300 +++ alsa-kernel/pci/hda/patch_realtek.c 2008-03-20 16:43:55.000000000 -0300 @@ -7650,6 +7650,7 @@ static struct snd_pci_quirk alc883_cfg_t SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720R), + SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720R), SND_PCI_QUIRK(0x1558, 0, "Clevo laptop", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), -- []'s Herton

1 0

[alsa-devel] pulseaudio plugin assertion on missing network link
by ChernYue Kwok 20 Mar '08

20 Mar '08

Hello, I am using Alsa with the Pulseaudio plugin on Fedora 8, with the RPMs listed below: alsa-utils-1.0.15-1.fc8 alsa-lib-devel-1.0.15-1.fc8 alsa-plugins-pulseaudio-1.0.15-2.fc8 alsa-lib-1.0.15-1.fc8 I am using the default configuration files from the alsa-plugins-pulseaudio RPM package. I have configured a static address on my server, this static address is set when my server boots. With my network cable plugged in, running 'amixer' returns the simple mixer controls on my server. However, when I run 'amixer' with my network cable unplugged, there is a long pause followed by the error message below: amixer: ctl_pulse.c:576: server_info_cb: Assertion 'ctl && i' failed Aborted Is there any way this assertion can be prevented? If so, how can I prevent this from happening? Thanks in advance -- Kwok Chern Yue

2 1

[alsa-devel] [PATCH] Use common 3stack-6ch mixer for 3stack-hp model
by Herton Ronaldo Krzesinski 20 Mar '08

20 Mar '08

Forgot one more: 3stack-hp model also have now the same mixer as 3stack-6ch (after DAC assignment fix in ALC883), so use it avoiding duplicating the same mixer definition. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com> diff -p -up alsa-kernel/pci/hda/patch_realtek.c.orig alsa-kernel/pci/hda/patch_realtek.c --- alsa-kernel/pci/hda/patch_realtek.c.orig 2008-03-19 14:23:56.000000000 -0300 +++ alsa-kernel/pci/hda/patch_realtek.c 2008-03-19 14:34:58.000000000 -0300 @@ -6954,44 +6954,6 @@ static struct snd_kcontrol_new alc883_me { } /* end */ }; -static struct snd_kcontrol_new alc888_3st_hp_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, 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_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = alc883_mux_enum_info, - .get = alc883_mux_enum_get, - .put = alc883_mux_enum_put, - }, - { } /* end */ -}; - static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -7889,9 +7851,9 @@ static struct alc_config_preset alc883_p .input_mux = &alc883_capture_source, .unsol_event = alc883_haier_w66_unsol_event, .init_hook = alc883_haier_w66_automute, - }, + }, [ALC888_3ST_HP] = { - .mixers = { alc888_3st_hp_mixer, alc883_chmode_mixer }, + .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc888_3st_hp_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, -- []'s Herton

2 1

[alsa-devel] 2.6.25-rc6-00074-g264e3e8: serious problem with ALSA Intel-HDA and MSI
by Prakash Punnoor 20 Mar '08

20 Mar '08

Hi, I didn't test an earlier 2.6.25-rc kernel, as I had no working nvidia driver... But the problem I am reporting happens before nvidia gets loaded. With 2.6.24.3 the ALSA driver didn't made any troubles for me. Stock kernel and enable_msi=1 made the driver to get a timeout and go back to legacy IRQ. With patches from http://kerneltrap.org/mailarchive/linux-kernel/2007/11/25/443612 2.6.24.x even works with MSI. Now with above 2.6.25-rc kernel regardless of using above patch, the kernel _hangs_ now when trying to initialize ALSA. It is even so bad, that doing a hw reset and trying to load patched 2.6.24 kernel causes ALSA to hang when trying to enable MSI. I need to turn power off and then patched 2.6.24 with MSI works again. 2.6.25-rc with ALSA Intel HDA _not_ forcing enable_msi causes it to boot. to summarize: 2.6.24 + patch + enable_msi=1: OK (and HDA MSI) 2.6.24 + enable_msi=1: OK (and HDA legacy IRQ) 2.6.24 + patch: OK (and HDA legacy IRQ) 2.6.24: OK (and HDA legacy IRQ) 2.6.25-rc + patch + enable_msi=1: hang 2.6.25-rc + enable_msi=1: hang 2.6.25-rc + patch: OK (and HDA legacy IRQ) 2.6.25-rc: OK (and HDA legacy IRQ) I don't know whether something in Linux MSI support was changed having effect on ALSA or whether the HDA driver has a regression. Is there a patch I can apply ontop of 2.6.24.3 to find out whether ALSA or possible MSI updates are the cause? I will be away from my desktop in 2 days for a month, so I cannot do a lot of testing nor bisecting. dmesg: 1 entries of 256 used Entering add_active_range(0, 1048576, 1245184) 2 entries of 256 used end_pfn_map = 1245184 DMI 2.4 present. ACPI: RSDP 000F7610, 0024 (r2 Nvidia) ACPI: XSDT CBEF3100, 004C (r1 Nvidia ASUSACPI 42302E31 AWRD 0) ACPI: FACP CBEF9C80, 00F4 (r3 Nvidia ASUSACPI 42302E31 AWRD 0) ACPI: DSDT CBEF3280, 6987 (r1 NVIDIA ASUSACPI 1000 MSFT 3000000) ACPI: FACS CBEF0000, 0040 ACPI: SSDT CBEF9E80, 01C4 (r1 PTLTD POWERNOW 1 LTP 1) ACPI: HPET CBEFA0C0, 0038 (r1 Nvidia ASUSACPI 42302E31 AWRD 98) ACPI: MCFG CBEFA140, 003C (r1 Nvidia ASUSACPI 42302E31 AWRD 0) ACPI: APIC CBEF9DC0, 007C (r1 Nvidia ASUSACPI 42302E31 AWRD 0) Entering add_active_range(0, 0, 159) 0 entries of 256 used Entering add_active_range(0, 256, 835312) 1 entries of 256 used Entering add_active_range(0, 1048576, 1245184) 2 entries of 256 used early res: 0 [0-fff] BIOS data page early res: 1 [6000-7fff] SMP_TRAMPOLINE early res: 2 [200000-7c4e2f] TEXT DATA BSS early res: 3 [9f000-9ffff] EBDA early res: 4 [8000-dfff] PGTABLE [ffffe20000000000-ffffe200001fffff] PMD ->ffff810001200000 on node 0 [ffffe20000200000-ffffe200003fffff] PMD ->ffff810001600000 on node 0 [ffffe20000400000-ffffe200005fffff] PMD ->ffff810001a00000 on node 0 [ffffe20000600000-ffffe200007fffff] PMD ->ffff810001e00000 on node 0 [ffffe20000800000-ffffe200009fffff] PMD ->ffff810002200000 on node 0 [ffffe20000a00000-ffffe20000bfffff] PMD ->ffff810002600000 on node 0 [ffffe20000c00000-ffffe20000dfffff] PMD ->ffff810002a00000 on node 0 [ffffe20000e00000-ffffe20000ffffff] PMD ->ffff810002e00000 on node 0 [ffffe20001000000-ffffe200011fffff] PMD ->ffff810003200000 on node 0 [ffffe20001200000-ffffe200013fffff] PMD ->ffff810003600000 on node 0 [ffffe20001400000-ffffe200015fffff] PMD ->ffff810003a00000 on node 0 [ffffe20001600000-ffffe200017fffff] PMD ->ffff810003e00000 on node 0 [ffffe20001800000-ffffe200019fffff] PMD ->ffff810004200000 on node 0 [ffffe20001a00000-ffffe20001bfffff] PMD ->ffff810004600000 on node 0 [ffffe20001c00000-ffffe20001dfffff] PMD ->ffff810004a00000 on node 0 [ffffe20001e00000-ffffe20001ffffff] PMD ->ffff810004e00000 on node 0 [ffffe20002000000-ffffe200021fffff] PMD ->ffff810005200000 on node 0 [ffffe20002200000-ffffe200023fffff] PMD ->ffff810005600000 on node 0 [ffffe20002400000-ffffe200025fffff] PMD ->ffff810005a00000 on node 0 [ffffe20002600000-ffffe200027fffff] PMD ->ffff810005e00000 on node 0 [ffffe20002800000-ffffe200029fffff] PMD ->ffff810006200000 on node 0 [ffffe20002a00000-ffffe20002bfffff] PMD ->ffff810006600000 on node 0 [ffffe20002c00000-ffffe20002dfffff] PMD ->ffff810006a00000 on node 0 [ffffe20002e00000-ffffe20002ffffff] PMD ->ffff810006e00000 on node 0 [ffffe20003000000-ffffe200031fffff] PMD ->ffff810007200000 on node 0 [ffffe20003200000-ffffe200033fffff] PMD ->ffff810007600000 on node 0 [ffffe20003400000-ffffe200035fffff] PMD ->ffff810007a00000 on node 0 [ffffe20003600000-ffffe200037fffff] PMD ->ffff810007e00000 on node 0 [ffffe20003800000-ffffe200039fffff] PMD ->ffff810008200000 on node 0 [ffffe20003a00000-ffffe20003bfffff] PMD ->ffff810008600000 on node 0 [ffffe20003c00000-ffffe20003dfffff] PMD ->ffff810008a00000 on node 0 [ffffe20003e00000-ffffe20003ffffff] PMD ->ffff810008e00000 on node 0 [ffffe20004000000-ffffe200041fffff] PMD ->ffff810009200000 on node 0 [ffffe20004200000-ffffe200043fffff] PMD ->ffff810009600000 on node 0 Zone PFN ranges: DMA 0 -> 4096 DMA32 4096 -> 1048576 Normal 1048576 -> 1245184 Movable zone start PFN for each node early_node_map[3] active PFN ranges 0: 0 -> 159 0: 256 -> 835312 0: 1048576 -> 1245184 On node 0 totalpages: 1031823 DMA zone: 56 pages used for memmap DMA zone: 1488 pages reserved DMA zone: 2455 pages, LIFO batch:0 DMA32 zone: 14280 pages used for memmap DMA32 zone: 816936 pages, LIFO batch:31 Normal zone: 2688 pages used for memmap Normal zone: 193920 pages, LIFO batch:31 Movable zone: 0 pages used for memmap ACPI: PM-Timer IO Port: 0x4008 ACPI: Local APIC address 0xfee00000 ACPI: LAPIC (acpi_id[0x00] lapic_id[0x00] enabled) Processor #0 (Bootup-CPU) ACPI: LAPIC (acpi_id[0x01] lapic_id[0x01] enabled) Processor #1 ACPI: LAPIC_NMI (acpi_id[0x00] high edge lint[0x1]) ACPI: LAPIC_NMI (acpi_id[0x01] high edge lint[0x1]) ACPI: IOAPIC (id[0x02] address[0xfec00000] gsi_base[0]) IOAPIC[0]: apic_id 2, address 0xfec00000, GSI 0-23 ACPI: INT_SRC_OVR (bus 0 bus_irq 0 global_irq 2 dfl dfl) ACPI: INT_SRC_OVR (bus 0 bus_irq 9 global_irq 9 high level) ACPI: INT_SRC_OVR (bus 0 bus_irq 14 global_irq 14 high edge) ACPI: INT_SRC_OVR (bus 0 bus_irq 15 global_irq 15 high edge) ACPI: IRQ0 used by override. ACPI: IRQ2 used by override. ACPI: IRQ9 used by override. ACPI: IRQ14 used by override. ACPI: IRQ15 used by override. Setting APIC routing to flat ACPI: HPET id: 0x10de8201 base: 0xfefff000 Using ACPI (MADT) for SMP configuration information Allocating PCI resources starting at d1000000 (gap: d0000000:10000000) SMP: Allowing 2 CPUs, 0 hotplug CPUs PERCPU: Allocating 33352 bytes of per cpu data Built 1 zonelists in Zone order, mobility grouping on. Total pages: 1013311 Kernel command line: root=/dev/md1 init=/sbin/init sata_nv.swncq=1 Initializing CPU#0 PID hash table entries: 4096 (order: 12, 32768 bytes) Extended CMOS year: 2000 TSC calibrated against PM_TIMER Marking TSC unstable due to TSCs unsynchronized time.c: Detected 2004.173 MHz processor. spurious 8259A interrupt: IRQ7. Console: colour VGA+ 80x25 console [tty0] enabled Dentry cache hash table entries: 524288 (order: 10, 4194304 bytes) Inode-cache hash table entries: 262144 (order: 9, 2097152 bytes) Checking aperture... Node 0: aperture @ c000000 size 32 MB Aperture pointing to e820 RAM. Ignoring. No AGP bridge found Your BIOS doesn't leave a aperture memory hole Please enable the IOMMU option in the BIOS setup This costs you 64 MB of RAM Mapping aperture over 65536 KB of RAM @ c000000 Memory: 3978932k/4980736k available (3684k kernel code, 147720k reserved, 1493k data, 256k init) CPA: page pool initialized 1 of 1 pages preallocated SLUB: Genslabs=12, HWalign=64, Order=0-1, MinObjects=4, CPUs=2, Nodes=1 hpet clockevent registered Calibrating delay using timer specific routine.. 4010.92 BogoMIPS (lpj=2005464) Mount-cache hash table entries: 256 CPU: L1 I Cache: 64K (64 bytes/line), D cache 64K (64 bytes/line) CPU: L2 Cache: 512K (64 bytes/line) CPU: Physical Processor ID: 0 CPU: Processor Core ID: 0 ACPI: Core revision 20070126 Using local APIC timer interrupts. APIC timer calibration result 12526078 Detected 12.526 MHz APIC timer. Booting processor 1/2 APIC 0x1 Initializing CPU#1 Calibrating delay using timer specific routine.. 4008.42 BogoMIPS (lpj=2004212) CPU: L1 I Cache: 64K (64 bytes/line), D cache 64K (64 bytes/line) CPU: L2 Cache: 512K (64 bytes/line) CPU: Physical Processor ID: 0 CPU: Processor Core ID: 1 AMD Athlon(tm) 64 X2 Dual Core Processor 3800+ stepping 02 Brought up 2 CPUs net_namespace: 872 bytes xor: automatically using best checksumming function: generic_sse generic_sse: 6100.000 MB/sec xor: using function: generic_sse (6100.000 MB/sec) NET: Registered protocol family 16 No dock devices found. ACPI: bus type pci registered PCI: Using MMCONFIG at e0000000 - efffffff PCI: Using configuration type 1 ACPI: EC: Look up EC in DSDT ACPI: Interpreter enabled ACPI: (supports S0 S1 S3 S5) ACPI: Using IOAPIC for interrupt routing ACPI: PCI Root Bridge [PCI0] (0000:00) PCI: Transparent bridge - 0000:00:10.0 ACPI: PCI Interrupt Routing Table [\_SB_.PCI0._PRT] ACPI: PCI Interrupt Routing Table [\_SB_.PCI0.HUB0._PRT] ACPI: PCI Interrupt Link [LNK1] (IRQs 5 7 9 10 11 14 15) *0, disabled. ACPI: PCI Interrupt Link [LNK2] (IRQs 5 7 9 10 11 14 15) *0, disabled. ACPI: PCI Interrupt Link [LNK3] (IRQs 5 7 9 10 11 14 15) *0, disabled. ACPI: PCI Interrupt Link [LNK4] (IRQs 5 7 9 10 11 14 15) *0, disabled. ACPI: PCI Interrupt Link [LNK5] (IRQs 5 7 9 10 *11 14 15) ACPI: PCI Interrupt Link [LNK6] (IRQs 5 7 9 10 11 14 15) *0, disabled. ACPI: PCI Interrupt Link [LNK7] (IRQs *5 7 9 10 11 14 15) ACPI: PCI Interrupt Link [LNK8] (IRQs 5 7 9 10 11 14 15) *0, disabled. ACPI: PCI Interrupt Link [LUBA] (IRQs 5 7 9 10 11 14 15) *0, disabled. ACPI: PCI Interrupt Link [LUBB] (IRQs 5 7 9 10 11 14 15) *0, disabled. ACPI: PCI Interrupt Link [LMAC] (IRQs 5 7 9 10 *11 14 15) ACPI: PCI Interrupt Link [LACI] (IRQs 5 7 9 10 11 14 15) *0, disabled. ACPI: PCI Interrupt Link [LAZA] (IRQs 5 7 9 10 *11 14 15) ACPI: PCI Interrupt Link [LPMU] (IRQs 5 7 9 10 11 14 15) *0, disabled. ACPI: PCI Interrupt Link [LMCI] (IRQs 5 7 9 10 11 14 15) *0, disabled. ACPI: PCI Interrupt Link [LSMB] (IRQs 5 7 9 10 11 14 15) *0, disabled. ACPI: PCI Interrupt Link [LUB2] (IRQs 5 7 9 10 11 14 15) *0, disabled. ACPI: PCI Interrupt Link [LIDE] (IRQs 5 7 9 10 11 14 15) *0, disabled. ACPI: PCI Interrupt Link [LSID] (IRQs *5 7 9 10 11 14 15) ACPI: PCI Interrupt Link [LFID] (IRQs *5 7 9 10 11 14 15) ACPI: PCI Interrupt Link [APC1] (IRQs 16) *0, disabled. ACPI: PCI Interrupt Link [APC2] (IRQs 17) *0, disabled. ACPI: PCI Interrupt Link [APC3] (IRQs 18) *0, disabled. ACPI: PCI Interrupt Link [APC4] (IRQs 19) *0, disabled. ACPI: PCI Interrupt Link [APC5] (IRQs 16) *0 ACPI: PCI Interrupt Link [APC6] (IRQs 16) *0, disabled. ACPI: PCI Interrupt Link [APC7] (IRQs 16) *0 ACPI: PCI Interrupt Link [APC8] (IRQs 16) *0, disabled. ACPI: PCI Interrupt Link [APCF] (IRQs 20 21 22 23) *0, disabled. ACPI: PCI Interrupt Link [APCG] (IRQs 20 21 22 23) *0, disabled. ACPI: PCI Interrupt Link [APCH] (IRQs 20 21 22 23) *0 ACPI: PCI Interrupt Link [APCJ] (IRQs 20 21 22 23) *0, disabled. ACPI: PCI Interrupt Link [APMU] (IRQs 20 21 22 23) *0, disabled. ACPI: PCI Interrupt Link [AAZA] (IRQs 20 21 22 23) *0 ACPI: PCI Interrupt Link [APCK] (IRQs 20 21 22 23) *0, disabled. ACPI: PCI Interrupt Link [APCS] (IRQs 20 21 22 23) *0, disabled. ACPI: PCI Interrupt Link [APCL] (IRQs 20 21 22 23) *0, disabled. ACPI: PCI Interrupt Link [APCM] (IRQs 20 21 22 23) *0, disabled. ACPI: PCI Interrupt Link [APCZ] (IRQs 20 21 22 23) *0, disabled. ACPI: PCI Interrupt Link [APSI] (IRQs 20 21 22 23) *0 ACPI: PCI Interrupt Link [APSJ] (IRQs 20 21 22 23) *0 Linux Plug and Play Support v0.97 (c) Adam Belay pnp: PnP ACPI init ACPI: bus type pnp registered pnpacpi: exceeded the max number of mem resources: 12 pnp: PnP ACPI: found 18 devices ACPI: ACPI bus type pnp unregistered SCSI subsystem initialized libata version 3.00 loaded. PCI: Using ACPI for IRQ routing PCI: If a device doesn't work, try "pci=routeirq". If it helps, post a report DMAR:parse DMAR table failure. PCI-DMA: Disabling AGP. PCI-DMA: aperture base @ c000000 size 65536 KB PCI-DMA: using GART IOMMU. PCI-DMA: Reserving 64MB of IOMMU area in the AGP aperture hpet0: at MMIO 0xfefff000, IRQs 2, 8, 31 hpet0: 3 32-bit timers, 25000000 Hz Time: hpet clocksource has been installed. Switched to high resolution mode on CPU 1 Switched to high resolution mode on CPU 0 system 00:01: ioport range 0x4000-0x407f has been reserved system 00:01: ioport range 0x4080-0x40ff has been reserved system 00:01: ioport range 0x4400-0x447f has been reserved system 00:01: ioport range 0x4480-0x44ff has been reserved system 00:01: ioport range 0x4800-0x487f has been reserved system 00:01: ioport range 0x4880-0x48ff has been reserved system 00:01: ioport range 0x2000-0x207f has been reserved system 00:01: ioport range 0x2080-0x20ff has been reserved system 00:01: iomem range 0xcc000000-0xcfffffff could not be reserved system 00:02: ioport range 0x4d0-0x4d1 has been reserved system 00:02: ioport range 0x800-0x87f has been reserved system 00:02: ioport range 0x290-0x297 has been reserved system 00:10: iomem range 0xe0000000-0xefffffff could not be reserved system 00:11: iomem range 0xf0000-0xf3fff could not be reserved system 00:11: iomem range 0xf4000-0xf7fff could not be reserved system 00:11: iomem range 0xf8000-0xfbfff could not be reserved system 00:11: iomem range 0xfc000-0xfffff could not be reserved system 00:11: iomem range 0xfefff000-0xfefff0ff has been reserved system 00:11: iomem range 0xcbef0000-0xcbefffff could not be reserved system 00:11: iomem range 0xffff0000-0xffffffff has been reserved system 00:11: iomem range 0x0-0x9ffff could not be reserved system 00:11: iomem range 0x100000-0xcbeeffff could not be reserved system 00:11: iomem range 0xcbf00000-0xcfefffff could not be reserved system 00:11: iomem range 0xfec00000-0xfec00fff has been reserved system 00:11: iomem range 0xfee00000-0xfeefffff could not be reserved PCI: Bridge: 0000:00:03.0 IO window: a000-afff MEM window: 0xfdc00000-0xfdcfffff PREFETCH window: 0x00000000fdd00000-0x00000000fddfffff PCI: Bridge: 0000:00:10.0 IO window: b000-bfff MEM window: 0xfdb00000-0xfdbfffff PREFETCH window: 0x00000000fde00000-0x00000000fdefffff PCI: Setting latency timer of device 0000:00:03.0 to 64 PCI: Setting latency timer of device 0000:00:10.0 to 64 NET: Registered protocol family 2 IP route cache hash table entries: 131072 (order: 8, 1048576 bytes) TCP established hash table entries: 262144 (order: 10, 4194304 bytes) TCP bind hash table entries: 65536 (order: 8, 1048576 bytes) TCP: Hash tables configured (established 262144 bind 65536) TCP reno registered Total HugeTLB memory allocated, 0 Installing knfsd (copyright (C) 1996 okir(a)monad.swb.de). async_tx: api initialized (sync-only) Block layer SCSI generic (bsg) driver version 0.4 loaded (major 254) io scheduler noop registered io scheduler anticipatory registered io scheduler deadline registered (default) io scheduler cfq registered pci 0000:00:00.0: Enabling HT MSI Mapping pci 0000:00:03.0: Enabling HT MSI Mapping pci 0000:00:05.0: Boot video device pci 0000:00:09.0: Enabling HT MSI Mapping pci 0000:00:0e.0: Enabling HT MSI Mapping pci 0000:00:0f.0: Enabling HT MSI Mapping pci 0000:00:10.0: Enabling HT MSI Mapping pci 0000:00:10.1: Enabling HT MSI Mapping PCI: Setting latency timer of device 0000:00:03.0 to 64 assign_interrupt_mode Found MSI capability PCI: 0000:00:00.0: enabled HT MSI mapping PCI: 0000:00:03.0: enabled HT MSI mapping Allocate Port Service[0000:00:03.0:pcie00] Allocate Port Service[0000:00:03.0:pcie03] input: Power Button (FF) as /class/input/input0 ACPI: Power Button (FF) [PWRF] input: Power Button (CM) as /class/input/input1 ACPI: Power Button (CM) [PWRB] ACPI: PNP0C0B:00 is registered as cooling_device0 ACPI: Fan [FAN] (on) ACPI: ACPI0007:00 is registered as cooling_device1 ACPI: ACPI0007:01 is registered as cooling_device2 ACPI: LNXTHERM:01 is registered as thermal_zone0 ACPI: Thermal Zone [THRM] (40 C) lp: driver loaded but no devices found Real Time Clock Driver v1.12ac hpet_resources: 0xfefff000 is busy Non-volatile memory driver v1.2 Linux agpgart interface v0.103 Serial: 8250/16550 driver $Revision: 1.90 $ 4 ports, IRQ sharing disabled serial8250: ttyS0 at I/O 0x3f8 (irq = 4) is a 16550A serial8250: ttyS1 at I/O 0x2f8 (irq = 3) is a 16550A 00:09: ttyS0 at I/O 0x3f8 (irq = 4) is a 16550A 00:0a: ttyS1 at I/O 0x2f8 (irq = 3) is a 16550A IT8716 SuperIO detected. parport_pc 00:0b: reported by Plug and Play ACPI parport0: PC-style at 0x378, irq 7 [PCSPP,TRISTATE,EPP] lp0: using parport0 (interrupt-driven). loop: module loaded tun: Universal TUN/TAP device driver, 1.6 tun: (C) 1999-2004 Max Krasnyansky <maxk(a)qualcomm.com> r8169 Gigabit Ethernet driver 2.2LK-NAPI loaded ACPI: PCI Interrupt Link [APC5] enabled at IRQ 16 ACPI: PCI Interrupt 0000:01:00.0[A] -> Link [APC5] -> GSI 16 (level, low) -> IRQ 16 PCI: Setting latency timer of device 0000:01:00.0 to 64 PCI: 0000:00:00.0: enabled HT MSI mapping PCI: 0000:00:03.0: enabled HT MSI mapping eth0: RTL8168b/8111b at 0xffffc20000026000, 00:08:54:52:28:41, XID 30000000 IRQ 318 Driver 'sd' needs updating - please use bus_type methods Driver 'sr' needs updating - please use bus_type methods sata_nv 0000:00:0e.0: version 3.5 ACPI: PCI Interrupt Link [APSI] enabled at IRQ 23 ACPI: PCI Interrupt 0000:00:0e.0[A] -> Link [APSI] -> GSI 23 (level, low) -> IRQ 23 sata_nv 0000:00:0e.0: Using SWNCQ mode PCI: Setting latency timer of device 0000:00:0e.0 to 64 scsi0 : sata_nv scsi1 : sata_nv ata1: SATA max UDMA/133 cmd 0x9f0 ctl 0xbf0 bmdma 0xe000 irq 23 ata2: SATA max UDMA/133 cmd 0x970 ctl 0xb70 bmdma 0xe008 irq 23 ata1: SATA link up 3.0 Gbps (SStatus 123 SControl 300) ata1.00: ATA-7: ST3320620AS, 3.AAK, max UDMA/133 ata1.00: 625142448 sectors, multi 1: LBA48 NCQ (depth 31/32) ata1.00: configured for UDMA/133 ata2: SATA link up 3.0 Gbps (SStatus 123 SControl 300) ata2.00: ATA-7: ST3320620AS, 3.AAK, max UDMA/133 ata2.00: 625142448 sectors, multi 1: LBA48 NCQ (depth 31/32) ata2.00: configured for UDMA/133 scsi 0:0:0:0: Direct-Access ATA ST3320620AS 3.AA PQ: 0 ANSI: 5 sd 0:0:0:0: [sda] 625142448 512-byte hardware sectors (320073 MB) sd 0:0:0:0: [sda] Write Protect is off sd 0:0:0:0: [sda] Mode Sense: 00 3a 00 00 sd 0:0:0:0: [sda] Write cache: enabled, read cache: enabled, doesn't support DPO or FUA sd 0:0:0:0: [sda] 625142448 512-byte hardware sectors (320073 MB) sd 0:0:0:0: [sda] Write Protect is off sd 0:0:0:0: [sda] Mode Sense: 00 3a 00 00 sd 0:0:0:0: [sda] Write cache: enabled, read cache: enabled, doesn't support DPO or FUA sda: sda1 sda2 < sda5 sda6 sda7 sda8 sda9 > sd 0:0:0:0: [sda] Attached SCSI disk sd 0:0:0:0: Attached scsi generic sg0 type 0 scsi 1:0:0:0: Direct-Access ATA ST3320620AS 3.AA PQ: 0 ANSI: 5 sd 1:0:0:0: [sdb] 625142448 512-byte hardware sectors (320073 MB) sd 1:0:0:0: [sdb] Write Protect is off sd 1:0:0:0: [sdb] Mode Sense: 00 3a 00 00 sd 1:0:0:0: [sdb] Write cache: enabled, read cache: enabled, doesn't support DPO or FUA sd 1:0:0:0: [sdb] 625142448 512-byte hardware sectors (320073 MB) sd 1:0:0:0: [sdb] Write Protect is off sd 1:0:0:0: [sdb] Mode Sense: 00 3a 00 00 sd 1:0:0:0: [sdb] Write cache: enabled, read cache: enabled, doesn't support DPO or FUA sdb: sdb1 sdb2 < sdb5 sdb6 sdb7 sdb8 sdb9 > sd 1:0:0:0: [sdb] Attached SCSI disk sd 1:0:0:0: Attached scsi generic sg1 type 0 ACPI: PCI Interrupt Link [APSJ] enabled at IRQ 22 ACPI: PCI Interrupt 0000:00:0f.0[A] -> Link [APSJ] -> GSI 22 (level, low) -> IRQ 22 sata_nv 0000:00:0f.0: Using SWNCQ mode PCI: Setting latency timer of device 0000:00:0f.0 to 64 scsi2 : sata_nv scsi3 : sata_nv ata3: SATA max UDMA/133 cmd 0x9e0 ctl 0xbe0 bmdma 0xcc00 irq 22 ata4: SATA max UDMA/133 cmd 0x960 ctl 0xb60 bmdma 0xcc08 irq 22 ata3: SATA link up 3.0 Gbps (SStatus 123 SControl 300) ata3.00: ATA-7: ST3320620AS, 3.AAK, max UDMA/133 ata3.00: 625142448 sectors, multi 1: LBA48 NCQ (depth 31/32) ata3.00: configured for UDMA/133 ata4: SATA link up 3.0 Gbps (SStatus 123 SControl 300) ata4.00: ATA-7: ST3320620AS, 3.AAD, max UDMA/133 ata4.00: 625142448 sectors, multi 1: LBA48 NCQ (depth 31/32) ata4.00: configured for UDMA/133 scsi 2:0:0:0: Direct-Access ATA ST3320620AS 3.AA PQ: 0 ANSI: 5 sd 2:0:0:0: [sdc] 625142448 512-byte hardware sectors (320073 MB) sd 2:0:0:0: [sdc] Write Protect is off sd 2:0:0:0: [sdc] Mode Sense: 00 3a 00 00 sd 2:0:0:0: [sdc] Write cache: enabled, read cache: enabled, doesn't support DPO or FUA sd 2:0:0:0: [sdc] 625142448 512-byte hardware sectors (320073 MB) sd 2:0:0:0: [sdc] Write Protect is off sd 2:0:0:0: [sdc] Mode Sense: 00 3a 00 00 sd 2:0:0:0: [sdc] Write cache: enabled, read cache: enabled, doesn't support DPO or FUA sdc: sdc1 sdc2 < sdc5 sdc6 sdc7 sdc8 sdc9 > sd 2:0:0:0: [sdc] Attached SCSI disk sd 2:0:0:0: Attached scsi generic sg2 type 0 scsi 3:0:0:0: Direct-Access ATA ST3320620AS 3.AA PQ: 0 ANSI: 5 sd 3:0:0:0: [sdd] 625142448 512-byte hardware sectors (320073 MB) sd 3:0:0:0: [sdd] Write Protect is off sd 3:0:0:0: [sdd] Mode Sense: 00 3a 00 00 sd 3:0:0:0: [sdd] Write cache: enabled, read cache: enabled, doesn't support DPO or FUA sd 3:0:0:0: [sdd] 625142448 512-byte hardware sectors (320073 MB) sd 3:0:0:0: [sdd] Write Protect is off sd 3:0:0:0: [sdd] Mode Sense: 00 3a 00 00 sd 3:0:0:0: [sdd] Write cache: enabled, read cache: enabled, doesn't support DPO or FUA sdd: sdd1 sdd2 < sdd5 sdd6 sdd7 sdd8 sdd9 > sd 3:0:0:0: [sdd] Attached SCSI disk sd 3:0:0:0: Attached scsi generic sg3 type 0 pata_amd 0000:00:0d.0: version 0.3.10 PCI: Setting latency timer of device 0000:00:0d.0 to 64 scsi4 : pata_amd scsi5 : pata_amd ata5: PATA max UDMA/133 cmd 0x1f0 ctl 0x3f6 bmdma 0xf400 irq 14 ata6: PATA max UDMA/133 cmd 0x170 ctl 0x376 bmdma 0xf408 irq 15 ata5.00: ATAPI: NU DVDRW DDW-081, BX32, max UDMA/33 ata5: nv_mode_filter: 0x739f&0x701f->0x701f, BIOS=0x7000 (0xc0000000) ACPI=0x701f (60:600:0x13) ata5.00: configured for UDMA/33 ata6: port disabled. ignoring. scsi 4:0:0:0: CD-ROM NU DVDRW DDW-081 BX32 PQ: 0 ANSI: 5 sr0: scsi3-mmc drive: 12x/40x writer cd/rw xa/form2 cdda tray Uniform CD-ROM driver Revision: 3.20 sr 4:0:0:0: Attached scsi CD-ROM sr0 sr 4:0:0:0: Attached scsi generic sg4 type 5 PNP: PS/2 Controller [PNP0303:PS2K,PNP0f13:PS2M] at 0x60,0x64 irq 1,12 serio: i8042 KBD port at 0x60,0x64 irq 1 serio: i8042 AUX port at 0x60,0x64 irq 12 gameport: NS558 PnP Gameport is pnp00:0f/gameport0, io 0x201, speed 637kHz mice: PS/2 mouse device common for all mice input: AT Translated Set 2 keyboard as /class/input/input2 i2c /dev entries driver i2c-adapter i2c-0: nForce2 SMBus adapter at 0x4c00 i2c-adapter i2c-1: nForce2 SMBus adapter at 0x4c40 it87: Found IT8716F chip at 0x290, revision 0 it87: in3 is VCC (+5V) it87: in7 is VCCH (+5V Stand-By) input: ImPS/2 Generic Wheel Mouse as /class/input/input3 md: raid0 personality registered for level 0 md: raid1 personality registered for level 1 raid6: int64x1 1996 MB/s raid6: int64x2 2460 MB/s raid6: int64x4 2601 MB/s raid6: int64x8 1980 MB/s raid6: sse2x1 2753 MB/s raid6: sse2x2 3695 MB/s raid6: sse2x4 3855 MB/s raid6: using algorithm sse2x4 (3855 MB/s) md: raid6 personality registered for level 6 md: raid5 personality registered for level 5 md: raid4 personality registered for level 4 md: multipath personality registered for level -4 device-mapper: uevent: version 1.0.3 device-mapper: ioctl: 4.13.0-ioctl (2007-10-18) initialised: dm-devel(a)redhat.com cpuidle: using governor ladder cpuidle: using governor menu Advanced Linux Sound Architecture Driver Version 1.0.16rc2 (Thu Jan 31 16:40:16 2008 UTC). ACPI: PCI Interrupt Link [AAZA] enabled at IRQ 21 ACPI: PCI Interrupt 0000:00:10.1[B] -> Link [AAZA] -> GSI 21 (level, low) -> IRQ 21 PCI: Setting latency timer of device 0000:00:10.1 to 64 ALSA device list: #0: HDA NVidia at 0xfe024000 irq 21 ip_tables: (C) 2000-2006 Netfilter Core Team TCP cubic registered NET: Registered protocol family 1 NET: Registered protocol family 17 RPC: Registered udp transport module. RPC: Registered tcp transport module. md: Autodetecting RAID arrays. md: Scanned 20 and added 20 devices. md: autorun ... md: considering sdd9 ... md: adding sdd9 ... md: sdd8 has different UUID to sdd9 md: sdd7 has different UUID to sdd9 md: sdd6 has different UUID to sdd9 md: sdd1 has different UUID to sdd9 md: adding sdc9 ... md: sdc8 has different UUID to sdd9 md: sdc7 has different UUID to sdd9 md: sdc6 has different UUID to sdd9 md: sdc1 has different UUID to sdd9 md: adding sdb9 ... md: sdb8 has different UUID to sdd9 md: sdb7 has different UUID to sdd9 md: sdb6 has different UUID to sdd9 md: sdb1 has different UUID to sdd9 md: adding sda9 ... md: sda8 has different UUID to sdd9 md: sda7 has different UUID to sdd9 md: sda6 has different UUID to sdd9 md: sda1 has different UUID to sdd9 md: created md4 md: bind<sda9> md: bind<sdb9> md: bind<sdc9> md: bind<sdd9> md: running: <sdd9><sdc9><sdb9><sda9> raid5: device sdd9 operational as raid disk 3 raid5: device sdc9 operational as raid disk 2 raid5: device sdb9 operational as raid disk 1 raid5: device sda9 operational as raid disk 0 raid5: allocated 4274kB for md4 raid5: raid level 5 set md4 active with 4 out of 4 devices, algorithm 2 RAID5 conf printout: --- rd:4 wd:4 disk 0, o:1, dev:sda9 disk 1, o:1, dev:sdb9 disk 2, o:1, dev:sdc9 disk 3, o:1, dev:sdd9 md: considering sdd8 ... md: adding sdd8 ... md: sdd7 has different UUID to sdd8 md: sdd6 has different UUID to sdd8 md: sdd1 has different UUID to sdd8 md: adding sdc8 ... md: sdc7 has different UUID to sdd8 md: sdc6 has different UUID to sdd8 md: sdc1 has different UUID to sdd8 md: adding sdb8 ... md: sdb7 has different UUID to sdd8 md: sdb6 has different UUID to sdd8 md: sdb1 has different UUID to sdd8 md: adding sda8 ... md: sda7 has different UUID to sdd8 md: sda6 has different UUID to sdd8 md: sda1 has different UUID to sdd8 md: created md3 md: bind<sda8> md: bind<sdb8> md: bind<sdc8> md: bind<sdd8> md: running: <sdd8><sdc8><sdb8><sda8> raid5: device sdd8 operational as raid disk 3 raid5: device sdc8 operational as raid disk 2 raid5: device sdb8 operational as raid disk 1 raid5: device sda8 operational as raid disk 0 raid5: allocated 4274kB for md3 raid5: raid level 5 set md3 active with 4 out of 4 devices, algorithm 2 RAID5 conf printout: --- rd:4 wd:4 disk 0, o:1, dev:sda8 disk 1, o:1, dev:sdb8 disk 2, o:1, dev:sdc8 disk 3, o:1, dev:sdd8 md: considering sdd7 ... md: adding sdd7 ... md: sdd6 has different UUID to sdd7 md: sdd1 has different UUID to sdd7 md: adding sdc7 ... md: sdc6 has different UUID to sdd7 md: sdc1 has different UUID to sdd7 md: adding sdb7 ... md: sdb6 has different UUID to sdd7 md: sdb1 has different UUID to sdd7 md: adding sda7 ... md: sda6 has different UUID to sdd7 md: sda1 has different UUID to sdd7 md: created md2 md: bind<sda7> md: bind<sdb7> md: bind<sdc7> md: bind<sdd7> md: running: <sdd7><sdc7><sdb7><sda7> md2: setting max_sectors to 128, segment boundary to 32767 raid0: looking at sdd7 raid0: comparing sdd7(2939776) with sdd7(2939776) raid0: END raid0: ==> UNIQUE raid0: 1 zones raid0: looking at sdc7 raid0: comparing sdc7(2939776) with sdd7(2939776) raid0: EQUAL raid0: looking at sdb7 raid0: comparing sdb7(2939776) with sdd7(2939776) raid0: EQUAL raid0: looking at sda7 raid0: comparing sda7(2939776) with sdd7(2939776) raid0: EQUAL raid0: FINAL 1 zones raid0: done. raid0 : md_size is 11759104 blocks. raid0 : conf->hash_spacing is 11759104 blocks. raid0 : nb_zone is 1. raid0 : Allocating 8 bytes for hash. md: considering sdd6 ... md: adding sdd6 ... md: sdd1 has different UUID to sdd6 md: adding sdc6 ... md: sdc1 has different UUID to sdd6 md: adding sdb6 ... md: sdb1 has different UUID to sdd6 md: adding sda6 ... md: sda1 has different UUID to sdd6 md: created md1 md: bind<sda6> md: bind<sdb6> md: bind<sdc6> md: bind<sdd6> md: running: <sdd6><sdc6><sdb6><sda6> raid5: device sdd6 operational as raid disk 3 raid5: device sdc6 operational as raid disk 2 raid5: device sdb6 operational as raid disk 1 raid5: device sda6 operational as raid disk 0 raid5: allocated 4274kB for md1 raid5: raid level 5 set md1 active with 4 out of 4 devices, algorithm 2 RAID5 conf printout: --- rd:4 wd:4 disk 0, o:1, dev:sda6 disk 1, o:1, dev:sdb6 disk 2, o:1, dev:sdc6 disk 3, o:1, dev:sdd6 md: considering sdd1 ... md: adding sdd1 ... md: adding sdc1 ... md: adding sdb1 ... md: adding sda1 ... md: created md0 md: bind<sda1> md: bind<sdb1> md: bind<sdc1> md: bind<sdd1> md: running: <sdd1><sdc1><sdb1><sda1> raid1: raid set md0 active with 4 out of 4 mirrors md: ... autorun DONE. EXT3-fs: mounted filesystem with writeback data mode. VFS: Mounted root (ext3 filesystem) readonly. Freeing unused kernel memory: 256k freed .config # # Automatically generated make config: don't edit # Linux kernel version: 2.6.25-rc6 # Wed Mar 19 20:19:54 2008 # CONFIG_64BIT=y # CONFIG_X86_32 is not set CONFIG_X86_64=y CONFIG_X86=y # CONFIG_GENERIC_LOCKBREAK is not set CONFIG_GENERIC_TIME=y CONFIG_GENERIC_CMOS_UPDATE=y CONFIG_CLOCKSOURCE_WATCHDOG=y CONFIG_GENERIC_CLOCKEVENTS=y CONFIG_GENERIC_CLOCKEVENTS_BROADCAST=y CONFIG_LOCKDEP_SUPPORT=y CONFIG_STACKTRACE_SUPPORT=y CONFIG_HAVE_LATENCYTOP_SUPPORT=y CONFIG_SEMAPHORE_SLEEPERS=y CONFIG_FAST_CMPXCHG_LOCAL=y CONFIG_MMU=y CONFIG_ZONE_DMA=y CONFIG_GENERIC_ISA_DMA=y CONFIG_GENERIC_IOMAP=y CONFIG_GENERIC_BUG=y CONFIG_GENERIC_HWEIGHT=y # CONFIG_GENERIC_GPIO is not set CONFIG_ARCH_MAY_HAVE_PC_FDC=y CONFIG_DMI=y CONFIG_RWSEM_GENERIC_SPINLOCK=y # CONFIG_RWSEM_XCHGADD_ALGORITHM is not set # CONFIG_ARCH_HAS_ILOG2_U32 is not set # CONFIG_ARCH_HAS_ILOG2_U64 is not set CONFIG_ARCH_HAS_CPU_IDLE_WAIT=y CONFIG_GENERIC_CALIBRATE_DELAY=y CONFIG_GENERIC_TIME_VSYSCALL=y CONFIG_ARCH_HAS_CPU_RELAX=y CONFIG_HAVE_SETUP_PER_CPU_AREA=y CONFIG_ARCH_HIBERNATION_POSSIBLE=y CONFIG_ARCH_SUSPEND_POSSIBLE=y CONFIG_ZONE_DMA32=y CONFIG_ARCH_POPULATES_NODE_MAP=y CONFIG_AUDIT_ARCH=y CONFIG_ARCH_SUPPORTS_AOUT=y CONFIG_GENERIC_HARDIRQS=y CONFIG_GENERIC_IRQ_PROBE=y CONFIG_GENERIC_PENDING_IRQ=y CONFIG_X86_SMP=y CONFIG_X86_64_SMP=y CONFIG_X86_TRAMPOLINE=y # CONFIG_KTIME_SCALAR is not set CONFIG_DEFCONFIG_LIST="/lib/modules/$UNAME_RELEASE/.config" # # General setup # CONFIG_EXPERIMENTAL=y CONFIG_LOCK_KERNEL=y CONFIG_INIT_ENV_ARG_LIMIT=32 CONFIG_LOCALVERSION="" CONFIG_LOCALVERSION_AUTO=y CONFIG_SWAP=y CONFIG_SYSVIPC=y CONFIG_SYSVIPC_SYSCTL=y CONFIG_POSIX_MQUEUE=y # CONFIG_BSD_PROCESS_ACCT is not set # CONFIG_TASKSTATS is not set # CONFIG_AUDIT is not set CONFIG_IKCONFIG=y CONFIG_IKCONFIG_PROC=y CONFIG_LOG_BUF_SHIFT=15 # CONFIG_CGROUPS is not set # CONFIG_GROUP_SCHED is not set CONFIG_SYSFS_DEPRECATED=y CONFIG_SYSFS_DEPRECATED_V2=y # CONFIG_RELAY is not set CONFIG_NAMESPACES=y # CONFIG_UTS_NS is not set # CONFIG_IPC_NS is not set # CONFIG_USER_NS is not set # CONFIG_PID_NS is not set # CONFIG_BLK_DEV_INITRD is not set # CONFIG_CC_OPTIMIZE_FOR_SIZE is not set CONFIG_SYSCTL=y # CONFIG_EMBEDDED is not set CONFIG_UID16=y CONFIG_SYSCTL_SYSCALL=y CONFIG_KALLSYMS=y # CONFIG_KALLSYMS_EXTRA_PASS is not set CONFIG_HOTPLUG=y CONFIG_PRINTK=y CONFIG_BUG=y CONFIG_ELF_CORE=y CONFIG_COMPAT_BRK=y CONFIG_BASE_FULL=y CONFIG_FUTEX=y CONFIG_ANON_INODES=y CONFIG_EPOLL=y CONFIG_SIGNALFD=y CONFIG_TIMERFD=y CONFIG_EVENTFD=y CONFIG_SHMEM=y CONFIG_VM_EVENT_COUNTERS=y CONFIG_SLUB_DEBUG=y # CONFIG_SLAB is not set CONFIG_SLUB=y # CONFIG_SLOB is not set CONFIG_PROFILING=y # CONFIG_MARKERS is not set CONFIG_OPROFILE=m CONFIG_HAVE_OPROFILE=y # CONFIG_KPROBES is not set CONFIG_HAVE_KPROBES=y CONFIG_HAVE_KRETPROBES=y CONFIG_PROC_PAGE_MONITOR=y CONFIG_SLABINFO=y CONFIG_RT_MUTEXES=y # CONFIG_TINY_SHMEM is not set CONFIG_BASE_SMALL=0 CONFIG_MODULES=y CONFIG_MODULE_UNLOAD=y CONFIG_MODULE_FORCE_UNLOAD=y # CONFIG_MODVERSIONS is not set # CONFIG_MODULE_SRCVERSION_ALL is not set CONFIG_KMOD=y CONFIG_STOP_MACHINE=y CONFIG_BLOCK=y # CONFIG_BLK_DEV_IO_TRACE is not set CONFIG_BLK_DEV_BSG=y CONFIG_BLOCK_COMPAT=y # # IO Schedulers # CONFIG_IOSCHED_NOOP=y CONFIG_IOSCHED_AS=y CONFIG_IOSCHED_DEADLINE=y CONFIG_IOSCHED_CFQ=y # CONFIG_DEFAULT_AS is not set CONFIG_DEFAULT_DEADLINE=y # CONFIG_DEFAULT_CFQ is not set # CONFIG_DEFAULT_NOOP is not set CONFIG_DEFAULT_IOSCHED="deadline" CONFIG_PREEMPT_NOTIFIERS=y CONFIG_CLASSIC_RCU=y # # Processor type and features # CONFIG_TICK_ONESHOT=y CONFIG_NO_HZ=y CONFIG_HIGH_RES_TIMERS=y CONFIG_GENERIC_CLOCKEVENTS_BUILD=y CONFIG_SMP=y CONFIG_X86_PC=y # CONFIG_X86_ELAN is not set # CONFIG_X86_VOYAGER is not set # CONFIG_X86_NUMAQ is not set # CONFIG_X86_SUMMIT is not set # CONFIG_X86_BIGSMP is not set # CONFIG_X86_VISWS is not set # CONFIG_X86_GENERICARCH is not set # CONFIG_X86_ES7000 is not set # CONFIG_X86_RDC321X is not set # CONFIG_X86_VSMP is not set # CONFIG_PARAVIRT_GUEST is not set # CONFIG_M386 is not set # CONFIG_M486 is not set # CONFIG_M586 is not set # CONFIG_M586TSC is not set # CONFIG_M586MMX is not set # CONFIG_M686 is not set # CONFIG_MPENTIUMII is not set # CONFIG_MPENTIUMIII is not set # CONFIG_MPENTIUMM is not set # CONFIG_MPENTIUM4 is not set # CONFIG_MK6 is not set # CONFIG_MK7 is not set CONFIG_MK8=y # CONFIG_MCRUSOE is not set # CONFIG_MEFFICEON is not set # CONFIG_MWINCHIPC6 is not set # CONFIG_MWINCHIP2 is not set # CONFIG_MWINCHIP3D is not set # CONFIG_MGEODEGX1 is not set # CONFIG_MGEODE_LX is not set # CONFIG_MCYRIXIII is not set # CONFIG_MVIAC3_2 is not set # CONFIG_MVIAC7 is not set # CONFIG_MPSC is not set # CONFIG_MCORE2 is not set # CONFIG_GENERIC_CPU is not set CONFIG_X86_L1_CACHE_BYTES=64 CONFIG_X86_INTERNODE_CACHE_BYTES=64 CONFIG_X86_CMPXCHG=y CONFIG_X86_L1_CACHE_SHIFT=6 CONFIG_X86_GOOD_APIC=y CONFIG_X86_INTEL_USERCOPY=y CONFIG_X86_USE_PPRO_CHECKSUM=y CONFIG_X86_TSC=y CONFIG_X86_MINIMUM_CPU_FAMILY=64 CONFIG_X86_DEBUGCTLMSR=y CONFIG_HPET_TIMER=y CONFIG_HPET_EMULATE_RTC=y CONFIG_GART_IOMMU=y # CONFIG_CALGARY_IOMMU is not set CONFIG_IOMMU_HELPER=y CONFIG_SWIOTLB=y CONFIG_NR_CPUS=2 # CONFIG_SCHED_SMT is not set CONFIG_SCHED_MC=y # CONFIG_PREEMPT_NONE is not set # CONFIG_PREEMPT_VOLUNTARY is not set CONFIG_PREEMPT=y # CONFIG_PREEMPT_RCU is not set CONFIG_X86_LOCAL_APIC=y CONFIG_X86_IO_APIC=y CONFIG_X86_MCE=y # CONFIG_X86_MCE_INTEL is not set CONFIG_X86_MCE_AMD=y # CONFIG_I8K is not set # CONFIG_MICROCODE is not set CONFIG_X86_MSR=y CONFIG_X86_CPUID=y # CONFIG_NUMA is not set CONFIG_ARCH_SPARSEMEM_DEFAULT=y CONFIG_ARCH_SPARSEMEM_ENABLE=y CONFIG_ARCH_SELECT_MEMORY_MODEL=y CONFIG_SELECT_MEMORY_MODEL=y # CONFIG_FLATMEM_MANUAL is not set # CONFIG_DISCONTIGMEM_MANUAL is not set CONFIG_SPARSEMEM_MANUAL=y CONFIG_SPARSEMEM=y CONFIG_HAVE_MEMORY_PRESENT=y # CONFIG_SPARSEMEM_STATIC is not set CONFIG_SPARSEMEM_EXTREME=y CONFIG_SPARSEMEM_VMEMMAP_ENABLE=y CONFIG_SPARSEMEM_VMEMMAP=y # CONFIG_MEMORY_HOTPLUG is not set CONFIG_SPLIT_PTLOCK_CPUS=4 CONFIG_RESOURCES_64BIT=y CONFIG_ZONE_DMA_FLAG=1 CONFIG_BOUNCE=y CONFIG_VIRT_TO_BUS=y CONFIG_MTRR=y # CONFIG_EFI is not set # CONFIG_SECCOMP is not set # CONFIG_HZ_100 is not set # CONFIG_HZ_250 is not set # CONFIG_HZ_300 is not set CONFIG_HZ_1000=y CONFIG_HZ=1000 CONFIG_SCHED_HRTICK=y CONFIG_KEXEC=y # CONFIG_CRASH_DUMP is not set CONFIG_PHYSICAL_START=0x200000 # CONFIG_RELOCATABLE is not set CONFIG_PHYSICAL_ALIGN=0x200000 CONFIG_HOTPLUG_CPU=y # CONFIG_COMPAT_VDSO is not set CONFIG_ARCH_ENABLE_MEMORY_HOTPLUG=y # # Power management options # CONFIG_PM=y # CONFIG_PM_LEGACY is not set # CONFIG_PM_DEBUG is not set CONFIG_PM_SLEEP_SMP=y CONFIG_PM_SLEEP=y CONFIG_SUSPEND=y CONFIG_SUSPEND_FREEZER=y # CONFIG_HIBERNATION is not set CONFIG_ACPI=y CONFIG_ACPI_SLEEP=y # CONFIG_ACPI_PROCFS is not set # CONFIG_ACPI_PROCFS_POWER is not set CONFIG_ACPI_SYSFS_POWER=y # CONFIG_ACPI_PROC_EVENT is not set # CONFIG_ACPI_AC is not set # CONFIG_ACPI_BATTERY is not set CONFIG_ACPI_BUTTON=y CONFIG_ACPI_FAN=y CONFIG_ACPI_DOCK=y # CONFIG_ACPI_BAY is not set CONFIG_ACPI_PROCESSOR=y CONFIG_ACPI_HOTPLUG_CPU=y CONFIG_ACPI_THERMAL=y # CONFIG_ACPI_WMI is not set # CONFIG_ACPI_ASUS is not set # CONFIG_ACPI_TOSHIBA is not set CONFIG_ACPI_CUSTOM_DSDT_FILE="" # CONFIG_ACPI_CUSTOM_DSDT is not set CONFIG_ACPI_BLACKLIST_YEAR=0 # CONFIG_ACPI_DEBUG is not set CONFIG_ACPI_EC=y CONFIG_ACPI_POWER=y CONFIG_ACPI_SYSTEM=y CONFIG_X86_PM_TIMER=y CONFIG_ACPI_CONTAINER=y # CONFIG_ACPI_SBS is not set # # CPU Frequency scaling # CONFIG_CPU_FREQ=y CONFIG_CPU_FREQ_TABLE=y # CONFIG_CPU_FREQ_DEBUG is not set CONFIG_CPU_FREQ_STAT=y # CONFIG_CPU_FREQ_STAT_DETAILS is not set # CONFIG_CPU_FREQ_DEFAULT_GOV_PERFORMANCE is not set # CONFIG_CPU_FREQ_DEFAULT_GOV_USERSPACE is not set CONFIG_CPU_FREQ_DEFAULT_GOV_ONDEMAND=y # CONFIG_CPU_FREQ_DEFAULT_GOV_CONSERVATIVE is not set CONFIG_CPU_FREQ_GOV_PERFORMANCE=y CONFIG_CPU_FREQ_GOV_POWERSAVE=y CONFIG_CPU_FREQ_GOV_USERSPACE=y CONFIG_CPU_FREQ_GOV_ONDEMAND=y CONFIG_CPU_FREQ_GOV_CONSERVATIVE=y # # CPUFreq processor drivers # # CONFIG_X86_ACPI_CPUFREQ is not set CONFIG_X86_POWERNOW_K8=m CONFIG_X86_POWERNOW_K8_ACPI=y # CONFIG_X86_SPEEDSTEP_CENTRINO is not set # CONFIG_X86_P4_CLOCKMOD is not set # # shared options # # CONFIG_X86_ACPI_CPUFREQ_PROC_INTF is not set # CONFIG_X86_SPEEDSTEP_LIB is not set CONFIG_CPU_IDLE=y CONFIG_CPU_IDLE_GOV_LADDER=y CONFIG_CPU_IDLE_GOV_MENU=y # # Bus options (PCI etc.) # CONFIG_PCI=y CONFIG_PCI_DIRECT=y CONFIG_PCI_MMCONFIG=y CONFIG_PCI_DOMAINS=y CONFIG_DMAR=y CONFIG_DMAR_GFX_WA=y CONFIG_DMAR_FLOPPY_WA=y CONFIG_PCIEPORTBUS=y CONFIG_PCIEAER=y CONFIG_ARCH_SUPPORTS_MSI=y CONFIG_PCI_MSI=y # CONFIG_PCI_LEGACY is not set CONFIG_HT_IRQ=y CONFIG_ISA_DMA_API=y CONFIG_K8_NB=y # CONFIG_PCCARD is not set # CONFIG_HOTPLUG_PCI is not set # # Executable file formats / Emulations # CONFIG_BINFMT_ELF=y CONFIG_COMPAT_BINFMT_ELF=y # CONFIG_BINFMT_MISC is not set CONFIG_IA32_EMULATION=y CONFIG_IA32_AOUT=y CONFIG_COMPAT=y CONFIG_COMPAT_FOR_U64_ALIGNMENT=y CONFIG_SYSVIPC_COMPAT=y # # Networking # CONFIG_NET=y # # Networking options # CONFIG_PACKET=y CONFIG_PACKET_MMAP=y CONFIG_UNIX=y CONFIG_XFRM=y # CONFIG_XFRM_USER is not set # CONFIG_XFRM_SUB_POLICY is not set # CONFIG_XFRM_MIGRATE is not set # CONFIG_XFRM_STATISTICS is not set # CONFIG_NET_KEY is not set CONFIG_INET=y CONFIG_IP_MULTICAST=y # CONFIG_IP_ADVANCED_ROUTER is not set CONFIG_IP_FIB_HASH=y # CONFIG_IP_PNP is not set # CONFIG_NET_IPIP is not set # CONFIG_NET_IPGRE is not set # CONFIG_IP_MROUTE is not set # CONFIG_ARPD is not set # CONFIG_SYN_COOKIES is not set # CONFIG_INET_AH is not set # CONFIG_INET_ESP is not set # CONFIG_INET_IPCOMP is not set # CONFIG_INET_XFRM_TUNNEL is not set # CONFIG_INET_TUNNEL is not set CONFIG_INET_XFRM_MODE_TRANSPORT=y CONFIG_INET_XFRM_MODE_TUNNEL=y CONFIG_INET_XFRM_MODE_BEET=y CONFIG_INET_LRO=y CONFIG_INET_DIAG=y CONFIG_INET_TCP_DIAG=y # CONFIG_TCP_CONG_ADVANCED is not set CONFIG_TCP_CONG_CUBIC=y CONFIG_DEFAULT_TCP_CONG="cubic" # CONFIG_TCP_MD5SIG is not set # CONFIG_IP_VS is not set # CONFIG_IPV6 is not set # CONFIG_INET6_XFRM_TUNNEL is not set # CONFIG_INET6_TUNNEL is not set # CONFIG_NETWORK_SECMARK is not set CONFIG_NETFILTER=y # CONFIG_NETFILTER_DEBUG is not set # CONFIG_NETFILTER_ADVANCED is not set # # Core Netfilter Configuration # # CONFIG_NETFILTER_NETLINK_LOG is not set # CONFIG_NF_CONNTRACK is not set CONFIG_NETFILTER_XTABLES=y CONFIG_NETFILTER_XT_TARGET_MARK=y CONFIG_NETFILTER_XT_TARGET_NFLOG=y CONFIG_NETFILTER_XT_TARGET_TCPMSS=y CONFIG_NETFILTER_XT_MATCH_MARK=y CONFIG_NETFILTER_XT_MATCH_POLICY=y # # IP: Netfilter Configuration # CONFIG_IP_NF_IPTABLES=y CONFIG_IP_NF_FILTER=y CONFIG_IP_NF_TARGET_REJECT=y CONFIG_IP_NF_TARGET_LOG=y CONFIG_IP_NF_TARGET_ULOG=y CONFIG_IP_NF_MANGLE=y # CONFIG_IP_DCCP is not set # CONFIG_IP_SCTP is not set # CONFIG_TIPC is not set # CONFIG_ATM is not set CONFIG_BRIDGE=y # CONFIG_VLAN_8021Q is not set # CONFIG_DECNET is not set CONFIG_LLC=y # CONFIG_LLC2 is not set # CONFIG_IPX is not set # CONFIG_ATALK is not set # CONFIG_X25 is not set # CONFIG_LAPB is not set # CONFIG_ECONET is not set # CONFIG_WAN_ROUTER is not set # CONFIG_NET_SCHED is not set # # Network testing # # CONFIG_NET_PKTGEN is not set # CONFIG_HAMRADIO is not set # CONFIG_CAN is not set # CONFIG_IRDA is not set # CONFIG_BT is not set # CONFIG_AF_RXRPC is not set # # Wireless # # CONFIG_CFG80211 is not set # CONFIG_WIRELESS_EXT is not set # CONFIG_MAC80211 is not set # CONFIG_IEEE80211 is not set # CONFIG_RFKILL is not set CONFIG_NET_9P=m CONFIG_NET_9P_FD=m # CONFIG_NET_9P_DEBUG is not set # # Device Drivers # # # Generic Driver Options # CONFIG_UEVENT_HELPER_PATH="/sbin/hotplug" # CONFIG_STANDALONE is not set CONFIG_PREVENT_FIRMWARE_BUILD=y CONFIG_FW_LOADER=y # CONFIG_SYS_HYPERVISOR is not set # CONFIG_CONNECTOR is not set # CONFIG_MTD is not set CONFIG_PARPORT=y CONFIG_PARPORT_PC=y CONFIG_PARPORT_SERIAL=y CONFIG_PARPORT_PC_FIFO=y CONFIG_PARPORT_PC_SUPERIO=y # CONFIG_PARPORT_GSC is not set # CONFIG_PARPORT_AX88796 is not set CONFIG_PARPORT_1284=y CONFIG_PNP=y # CONFIG_PNP_DEBUG is not set # # Protocols # CONFIG_PNPACPI=y CONFIG_BLK_DEV=y # CONFIG_BLK_DEV_FD is not set # CONFIG_PARIDE is not set # CONFIG_BLK_CPQ_DA is not set # CONFIG_BLK_CPQ_CISS_DA is not set # CONFIG_BLK_DEV_DAC960 is not set # CONFIG_BLK_DEV_UMEM is not set # CONFIG_BLK_DEV_COW_COMMON is not set CONFIG_BLK_DEV_LOOP=y # CONFIG_BLK_DEV_CRYPTOLOOP is not set # CONFIG_BLK_DEV_NBD is not set # CONFIG_BLK_DEV_SX8 is not set # CONFIG_BLK_DEV_UB is not set # CONFIG_BLK_DEV_RAM is not set # CONFIG_CDROM_PKTCDVD is not set # CONFIG_ATA_OVER_ETH is not set # CONFIG_MISC_DEVICES is not set CONFIG_HAVE_IDE=y # CONFIG_IDE is not set # # SCSI device support # # CONFIG_RAID_ATTRS is not set CONFIG_SCSI=y CONFIG_SCSI_DMA=y # CONFIG_SCSI_TGT is not set # CONFIG_SCSI_NETLINK is not set # CONFIG_SCSI_PROC_FS is not set # # SCSI support type (disk, tape, CD-ROM) # CONFIG_BLK_DEV_SD=y # CONFIG_CHR_DEV_ST is not set # CONFIG_CHR_DEV_OSST is not set CONFIG_BLK_DEV_SR=y # CONFIG_BLK_DEV_SR_VENDOR is not set CONFIG_CHR_DEV_SG=y # CONFIG_CHR_DEV_SCH is not set # # Some SCSI devices (e.g. CD jukebox) support multiple LUNs # # CONFIG_SCSI_MULTI_LUN is not set # CONFIG_SCSI_CONSTANTS is not set # CONFIG_SCSI_LOGGING is not set # CONFIG_SCSI_SCAN_ASYNC is not set CONFIG_SCSI_WAIT_SCAN=m # # SCSI Transports # # CONFIG_SCSI_SPI_ATTRS is not set # CONFIG_SCSI_FC_ATTRS is not set # CONFIG_SCSI_ISCSI_ATTRS is not set # CONFIG_SCSI_SAS_ATTRS is not set # CONFIG_SCSI_SAS_LIBSAS is not set # CONFIG_SCSI_SRP_ATTRS is not set # CONFIG_SCSI_LOWLEVEL is not set CONFIG_ATA=y # CONFIG_ATA_NONSTANDARD is not set CONFIG_ATA_ACPI=y # CONFIG_SATA_AHCI is not set # CONFIG_SATA_SVW is not set # CONFIG_ATA_PIIX is not set # CONFIG_SATA_MV is not set CONFIG_SATA_NV=y # CONFIG_PDC_ADMA is not set # CONFIG_SATA_QSTOR is not set # CONFIG_SATA_PROMISE is not set # CONFIG_SATA_SX4 is not set # CONFIG_SATA_SIL is not set # CONFIG_SATA_SIL24 is not set # CONFIG_SATA_SIS is not set # CONFIG_SATA_ULI is not set # CONFIG_SATA_VIA is not set # CONFIG_SATA_VITESSE is not set # CONFIG_SATA_INIC162X is not set CONFIG_PATA_ACPI=y # CONFIG_PATA_ALI is not set CONFIG_PATA_AMD=y # CONFIG_PATA_ARTOP is not set # CONFIG_PATA_ATIIXP is not set # CONFIG_PATA_CMD640_PCI is not set # CONFIG_PATA_CMD64X is not set # CONFIG_PATA_CS5520 is not set # CONFIG_PATA_CS5530 is not set # CONFIG_PATA_CYPRESS is not set # CONFIG_PATA_EFAR is not set # CONFIG_ATA_GENERIC is not set # CONFIG_PATA_HPT366 is not set # CONFIG_PATA_HPT37X is not set # CONFIG_PATA_HPT3X2N is not set # CONFIG_PATA_HPT3X3 is not set # CONFIG_PATA_IT821X is not set # CONFIG_PATA_IT8213 is not set # CONFIG_PATA_JMICRON is not set # CONFIG_PATA_TRIFLEX is not set # CONFIG_PATA_MARVELL is not set # CONFIG_PATA_MPIIX is not set # CONFIG_PATA_OLDPIIX is not set # CONFIG_PATA_NETCELL is not set # CONFIG_PATA_NINJA32 is not set # CONFIG_PATA_NS87410 is not set # CONFIG_PATA_NS87415 is not set # CONFIG_PATA_OPTI is not set # CONFIG_PATA_OPTIDMA is not set # CONFIG_PATA_PDC_OLD is not set # CONFIG_PATA_RADISYS is not set # CONFIG_PATA_RZ1000 is not set # CONFIG_PATA_SC1200 is not set # CONFIG_PATA_SERVERWORKS is not set # CONFIG_PATA_PDC2027X is not set # CONFIG_PATA_SIL680 is not set # CONFIG_PATA_SIS is not set # CONFIG_PATA_VIA is not set # CONFIG_PATA_WINBOND is not set CONFIG_MD=y CONFIG_BLK_DEV_MD=y # CONFIG_MD_LINEAR is not set CONFIG_MD_RAID0=y CONFIG_MD_RAID1=y # CONFIG_MD_RAID10 is not set CONFIG_MD_RAID456=y CONFIG_MD_RAID5_RESHAPE=y CONFIG_MD_MULTIPATH=y # CONFIG_MD_FAULTY is not set CONFIG_BLK_DEV_DM=y # CONFIG_DM_DEBUG is not set # CONFIG_DM_CRYPT is not set # CONFIG_DM_SNAPSHOT is not set # CONFIG_DM_MIRROR is not set # CONFIG_DM_ZERO is not set # CONFIG_DM_MULTIPATH is not set # CONFIG_DM_DELAY is not set CONFIG_DM_UEVENT=y # CONFIG_FUSION is not set # # IEEE 1394 (FireWire) support # # CONFIG_FIREWIRE is not set # CONFIG_IEEE1394 is not set # CONFIG_I2O is not set # CONFIG_MACINTOSH_DRIVERS is not set CONFIG_NETDEVICES=y CONFIG_NETDEVICES_MULTIQUEUE=y # CONFIG_DUMMY is not set # CONFIG_BONDING is not set # CONFIG_MACVLAN is not set # CONFIG_EQUALIZER is not set CONFIG_TUN=y # CONFIG_VETH is not set # CONFIG_NET_SB1000 is not set # CONFIG_ARCNET is not set # CONFIG_PHYLIB is not set CONFIG_NET_ETHERNET=y CONFIG_MII=y # CONFIG_HAPPYMEAL is not set # CONFIG_SUNGEM is not set # CONFIG_CASSINI is not set # CONFIG_NET_VENDOR_3COM is not set # CONFIG_NET_TULIP is not set # CONFIG_HP100 is not set # CONFIG_IBM_NEW_EMAC_ZMII is not set # CONFIG_IBM_NEW_EMAC_RGMII is not set # CONFIG_IBM_NEW_EMAC_TAH is not set # CONFIG_IBM_NEW_EMAC_EMAC4 is not set CONFIG_NET_PCI=y # CONFIG_PCNET32 is not set # CONFIG_AMD8111_ETH is not set # CONFIG_ADAPTEC_STARFIRE is not set # CONFIG_B44 is not set CONFIG_FORCEDETH=m CONFIG_FORCEDETH_NAPI=y # CONFIG_EEPRO100 is not set # CONFIG_E100 is not set # CONFIG_FEALNX is not set # CONFIG_NATSEMI is not set # CONFIG_NE2K_PCI is not set # CONFIG_8139CP is not set # CONFIG_8139TOO is not set # CONFIG_R6040 is not set # CONFIG_SIS900 is not set # CONFIG_EPIC100 is not set # CONFIG_SUNDANCE is not set # CONFIG_VIA_RHINE is not set # CONFIG_SC92031 is not set # CONFIG_NET_POCKET is not set CONFIG_NETDEV_1000=y # CONFIG_ACENIC is not set # CONFIG_DL2K is not set # CONFIG_E1000 is not set # CONFIG_E1000E is not set # CONFIG_E1000E_ENABLED is not set # CONFIG_IP1000 is not set # CONFIG_IGB is not set # CONFIG_NS83820 is not set # CONFIG_HAMACHI is not set # CONFIG_YELLOWFIN is not set CONFIG_R8169=y CONFIG_R8169_NAPI=y # CONFIG_SIS190 is not set # CONFIG_SKGE is not set # CONFIG_SKY2 is not set # CONFIG_SK98LIN is not set # CONFIG_VIA_VELOCITY is not set # CONFIG_TIGON3 is not set # CONFIG_BNX2 is not set # CONFIG_QLA3XXX is not set # CONFIG_ATL1 is not set # CONFIG_NETDEV_10000 is not set # CONFIG_TR is not set # # Wireless LAN # # CONFIG_WLAN_PRE80211 is not set # CONFIG_WLAN_80211 is not set # # USB Network Adapters # # CONFIG_USB_CATC is not set # CONFIG_USB_KAWETH is not set # CONFIG_USB_PEGASUS is not set # CONFIG_USB_RTL8150 is not set # CONFIG_USB_USBNET is not set # CONFIG_WAN is not set # CONFIG_FDDI is not set # CONFIG_HIPPI is not set # CONFIG_PLIP is not set # CONFIG_PPP is not set # CONFIG_SLIP is not set # CONFIG_NET_FC is not set # CONFIG_NETCONSOLE is not set # CONFIG_NETPOLL is not set # CONFIG_NET_POLL_CONTROLLER is not set # CONFIG_ISDN is not set # CONFIG_PHONE is not set # # Input device support # CONFIG_INPUT=y # CONFIG_INPUT_FF_MEMLESS is not set # CONFIG_INPUT_POLLDEV is not set # # Userland interfaces # CONFIG_INPUT_MOUSEDEV=y # CONFIG_INPUT_MOUSEDEV_PSAUX is not set CONFIG_INPUT_MOUSEDEV_SCREEN_X=1024 CONFIG_INPUT_MOUSEDEV_SCREEN_Y=768 # CONFIG_INPUT_JOYDEV is not set CONFIG_INPUT_EVDEV=y # CONFIG_INPUT_EVBUG is not set # # Input Device Drivers # CONFIG_INPUT_KEYBOARD=y CONFIG_KEYBOARD_ATKBD=y # CONFIG_KEYBOARD_SUNKBD is not set # CONFIG_KEYBOARD_LKKBD is not set # CONFIG_KEYBOARD_XTKBD is not set # CONFIG_KEYBOARD_NEWTON is not set # CONFIG_KEYBOARD_STOWAWAY is not set CONFIG_INPUT_MOUSE=y CONFIG_MOUSE_PS2=y CONFIG_MOUSE_PS2_ALPS=y CONFIG_MOUSE_PS2_LOGIPS2PP=y CONFIG_MOUSE_PS2_SYNAPTICS=y CONFIG_MOUSE_PS2_LIFEBOOK=y CONFIG_MOUSE_PS2_TRACKPOINT=y # CONFIG_MOUSE_PS2_TOUCHKIT is not set # CONFIG_MOUSE_SERIAL is not set # CONFIG_MOUSE_APPLETOUCH is not set # CONFIG_MOUSE_VSXXXAA is not set # CONFIG_INPUT_JOYSTICK is not set # CONFIG_INPUT_TABLET is not set # CONFIG_INPUT_TOUCHSCREEN is not set # CONFIG_INPUT_MISC is not set # # Hardware I/O ports # CONFIG_SERIO=y CONFIG_SERIO_I8042=y # CONFIG_SERIO_SERPORT is not set # CONFIG_SERIO_CT82C710 is not set # CONFIG_SERIO_PARKBD is not set # CONFIG_SERIO_PCIPS2 is not set CONFIG_SERIO_LIBPS2=y # CONFIG_SERIO_RAW is not set CONFIG_GAMEPORT=y CONFIG_GAMEPORT_NS558=y # CONFIG_GAMEPORT_L4 is not set # CONFIG_GAMEPORT_EMU10K1 is not set # CONFIG_GAMEPORT_FM801 is not set # # Character devices # CONFIG_VT=y CONFIG_VT_CONSOLE=y CONFIG_HW_CONSOLE=y # CONFIG_VT_HW_CONSOLE_BINDING is not set # CONFIG_SERIAL_NONSTANDARD is not set # CONFIG_NOZOMI is not set # # Serial drivers # CONFIG_SERIAL_8250=y CONFIG_SERIAL_8250_CONSOLE=y CONFIG_FIX_EARLYCON_MEM=y CONFIG_SERIAL_8250_PCI=y CONFIG_SERIAL_8250_PNP=y CONFIG_SERIAL_8250_NR_UARTS=4 CONFIG_SERIAL_8250_RUNTIME_UARTS=4 # CONFIG_SERIAL_8250_EXTENDED is not set # # Non-8250 serial port support # CONFIG_SERIAL_CORE=y CONFIG_SERIAL_CORE_CONSOLE=y # CONFIG_SERIAL_JSM is not set CONFIG_UNIX98_PTYS=y # CONFIG_LEGACY_PTYS is not set CONFIG_PRINTER=y # CONFIG_LP_CONSOLE is not set # CONFIG_PPDEV is not set # CONFIG_IPMI_HANDLER is not set # CONFIG_HW_RANDOM is not set CONFIG_NVRAM=y CONFIG_RTC=y # CONFIG_R3964 is not set # CONFIG_APPLICOM is not set # CONFIG_MWAVE is not set # CONFIG_PC8736x_GPIO is not set # CONFIG_RAW_DRIVER is not set CONFIG_HPET=y # CONFIG_HPET_RTC_IRQ is not set CONFIG_HPET_MMAP=y # CONFIG_HANGCHECK_TIMER is not set # CONFIG_TCG_TPM is not set # CONFIG_TELCLOCK is not set CONFIG_DEVPORT=y CONFIG_I2C=y CONFIG_I2C_BOARDINFO=y CONFIG_I2C_CHARDEV=y # # I2C Algorithms # # CONFIG_I2C_ALGOBIT is not set # CONFIG_I2C_ALGOPCF is not set # CONFIG_I2C_ALGOPCA is not set # # I2C Hardware Bus support # # CONFIG_I2C_ALI1535 is not set # CONFIG_I2C_ALI1563 is not set # CONFIG_I2C_ALI15X3 is not set # CONFIG_I2C_AMD756 is not set # CONFIG_I2C_AMD8111 is not set # CONFIG_I2C_I801 is not set # CONFIG_I2C_I810 is not set # CONFIG_I2C_PIIX4 is not set CONFIG_I2C_NFORCE2=y # CONFIG_I2C_OCORES is not set # CONFIG_I2C_PARPORT is not set # CONFIG_I2C_PARPORT_LIGHT is not set # CONFIG_I2C_PROSAVAGE is not set # CONFIG_I2C_SAVAGE4 is not set # CONFIG_I2C_SIMTEC is not set # CONFIG_I2C_SIS5595 is not set # CONFIG_I2C_SIS630 is not set # CONFIG_I2C_SIS96X is not set # CONFIG_I2C_TAOS_EVM is not set # CONFIG_I2C_STUB is not set # CONFIG_I2C_TINY_USB is not set # CONFIG_I2C_VIA is not set # CONFIG_I2C_VIAPRO is not set # CONFIG_I2C_VOODOO3 is not set # # Miscellaneous I2C Chip support # # CONFIG_DS1682 is not set # CONFIG_SENSORS_EEPROM is not set # CONFIG_SENSORS_PCF8574 is not set # CONFIG_PCF8575 is not set # CONFIG_SENSORS_PCF8591 is not set # CONFIG_TPS65010 is not set # CONFIG_SENSORS_MAX6875 is not set # CONFIG_SENSORS_TSL2550 is not set # CONFIG_I2C_DEBUG_CORE is not set # CONFIG_I2C_DEBUG_ALGO is not set # CONFIG_I2C_DEBUG_BUS is not set # CONFIG_I2C_DEBUG_CHIP is not set # # SPI support # # CONFIG_SPI is not set # CONFIG_SPI_MASTER is not set # CONFIG_W1 is not set CONFIG_POWER_SUPPLY=y # CONFIG_POWER_SUPPLY_DEBUG is not set # CONFIG_PDA_POWER is not set # CONFIG_BATTERY_DS2760 is not set CONFIG_HWMON=y CONFIG_HWMON_VID=y # CONFIG_SENSORS_ABITUGURU is not set # CONFIG_SENSORS_ABITUGURU3 is not set # CONFIG_SENSORS_AD7418 is not set # CONFIG_SENSORS_ADM1021 is not set # CONFIG_SENSORS_ADM1025 is not set # CONFIG_SENSORS_ADM1026 is not set # CONFIG_SENSORS_ADM1029 is not set # CONFIG_SENSORS_ADM1031 is not set # CONFIG_SENSORS_ADM9240 is not set # CONFIG_SENSORS_ADT7470 is not set # CONFIG_SENSORS_ADT7473 is not set CONFIG_SENSORS_K8TEMP=y # CONFIG_SENSORS_ASB100 is not set # CONFIG_SENSORS_ATXP1 is not set # CONFIG_SENSORS_DS1621 is not set # CONFIG_SENSORS_I5K_AMB is not set # CONFIG_SENSORS_F71805F is not set # CONFIG_SENSORS_F71882FG is not set # CONFIG_SENSORS_F75375S is not set # CONFIG_SENSORS_FSCHER is not set # CONFIG_SENSORS_FSCPOS is not set # CONFIG_SENSORS_FSCHMD is not set # CONFIG_SENSORS_GL518SM is not set # CONFIG_SENSORS_GL520SM is not set # CONFIG_SENSORS_CORETEMP is not set CONFIG_SENSORS_IT87=y # CONFIG_SENSORS_LM63 is not set # CONFIG_SENSORS_LM75 is not set # CONFIG_SENSORS_LM77 is not set # CONFIG_SENSORS_LM78 is not set # CONFIG_SENSORS_LM80 is not set # CONFIG_SENSORS_LM83 is not set # CONFIG_SENSORS_LM85 is not set # CONFIG_SENSORS_LM87 is not set # CONFIG_SENSORS_LM90 is not set # CONFIG_SENSORS_LM92 is not set # CONFIG_SENSORS_LM93 is not set # CONFIG_SENSORS_MAX1619 is not set # CONFIG_SENSORS_MAX6650 is not set # CONFIG_SENSORS_PC87360 is not set # CONFIG_SENSORS_PC87427 is not set # CONFIG_SENSORS_SIS5595 is not set # CONFIG_SENSORS_DME1737 is not set # CONFIG_SENSORS_SMSC47M1 is not set # CONFIG_SENSORS_SMSC47M192 is not set # CONFIG_SENSORS_SMSC47B397 is not set # CONFIG_SENSORS_ADS7828 is not set # CONFIG_SENSORS_THMC50 is not set # CONFIG_SENSORS_VIA686A is not set # CONFIG_SENSORS_VT1211 is not set # CONFIG_SENSORS_VT8231 is not set # CONFIG_SENSORS_W83781D is not set # CONFIG_SENSORS_W83791D is not set # CONFIG_SENSORS_W83792D is not set # CONFIG_SENSORS_W83793 is not set # CONFIG_SENSORS_W83L785TS is not set # CONFIG_SENSORS_W83L786NG is not set # CONFIG_SENSORS_W83627HF is not set # CONFIG_SENSORS_W83627EHF is not set # CONFIG_SENSORS_HDAPS is not set # CONFIG_SENSORS_APPLESMC is not set # CONFIG_HWMON_DEBUG_CHIP is not set CONFIG_THERMAL=y # CONFIG_WATCHDOG is not set # # Sonics Silicon Backplane # CONFIG_SSB_POSSIBLE=y # CONFIG_SSB is not set # # Multifunction device drivers # # CONFIG_MFD_SM501 is not set # # Multimedia devices # # CONFIG_VIDEO_DEV is not set # CONFIG_DVB_CORE is not set # CONFIG_DAB is not set # # Graphics support # CONFIG_AGP=y CONFIG_AGP_AMD64=y # CONFIG_AGP_INTEL is not set # CONFIG_AGP_SIS is not set # CONFIG_AGP_VIA is not set # CONFIG_DRM is not set # CONFIG_VGASTATE is not set # CONFIG_VIDEO_OUTPUT_CONTROL is not set # CONFIG_FB is not set # CONFIG_BACKLIGHT_LCD_SUPPORT is not set # # Display device support # # CONFIG_DISPLAY_SUPPORT is not set # # Console display driver support # CONFIG_VGA_CONSOLE=y CONFIG_VGACON_SOFT_SCROLLBACK=y CONFIG_VGACON_SOFT_SCROLLBACK_SIZE=64 CONFIG_VIDEO_SELECT=y CONFIG_DUMMY_CONSOLE=y # # Sound # CONFIG_SOUND=y # # Advanced Linux Sound Architecture # CONFIG_SND=y CONFIG_SND_TIMER=y CONFIG_SND_PCM=y # CONFIG_SND_SEQUENCER is not set CONFIG_SND_OSSEMUL=y CONFIG_SND_MIXER_OSS=y CONFIG_SND_PCM_OSS=y CONFIG_SND_PCM_OSS_PLUGINS=y CONFIG_SND_RTCTIMER=y CONFIG_SND_DYNAMIC_MINORS=y CONFIG_SND_SUPPORT_OLD_API=y # CONFIG_SND_VERBOSE_PROCFS is not set # CONFIG_SND_VERBOSE_PRINTK is not set # CONFIG_SND_DEBUG is not set # # Generic devices # # CONFIG_SND_DUMMY is not set # CONFIG_SND_MTPAV is not set # CONFIG_SND_MTS64 is not set # CONFIG_SND_SERIAL_U16550 is not set # CONFIG_SND_MPU401 is not set # CONFIG_SND_PORTMAN2X4 is not set # # PCI devices # # CONFIG_SND_AD1889 is not set # CONFIG_SND_ALS300 is not set # CONFIG_SND_ALS4000 is not set # CONFIG_SND_ALI5451 is not set # CONFIG_SND_ATIIXP is not set # CONFIG_SND_ATIIXP_MODEM is not set # CONFIG_SND_AU8810 is not set # CONFIG_SND_AU8820 is not set # CONFIG_SND_AU8830 is not set # CONFIG_SND_AZT3328 is not set # CONFIG_SND_BT87X is not set # CONFIG_SND_CA0106 is not set # CONFIG_SND_CMIPCI is not set # CONFIG_SND_OXYGEN is not set # CONFIG_SND_CS4281 is not set # CONFIG_SND_CS46XX is not set # CONFIG_SND_CS5530 is not set # CONFIG_SND_DARLA20 is not set # CONFIG_SND_GINA20 is not set # CONFIG_SND_LAYLA20 is not set # CONFIG_SND_DARLA24 is not set # CONFIG_SND_GINA24 is not set # CONFIG_SND_LAYLA24 is not set # CONFIG_SND_MONA is not set # CONFIG_SND_MIA is not set # CONFIG_SND_ECHO3G is not set # CONFIG_SND_INDIGO is not set # CONFIG_SND_INDIGOIO is not set # CONFIG_SND_INDIGODJ is not set # CONFIG_SND_EMU10K1 is not set # CONFIG_SND_EMU10K1X is not set # CONFIG_SND_ENS1370 is not set # CONFIG_SND_ENS1371 is not set # CONFIG_SND_ES1938 is not set # CONFIG_SND_ES1968 is not set # CONFIG_SND_FM801 is not set CONFIG_SND_HDA_INTEL=y # CONFIG_SND_HDA_HWDEP is not set # CONFIG_SND_HDA_CODEC_REALTEK is not set CONFIG_SND_HDA_CODEC_ANALOG=y # CONFIG_SND_HDA_CODEC_SIGMATEL is not set # CONFIG_SND_HDA_CODEC_VIA is not set # CONFIG_SND_HDA_CODEC_ATIHDMI is not set # CONFIG_SND_HDA_CODEC_CONEXANT is not set # CONFIG_SND_HDA_CODEC_CMEDIA is not set # CONFIG_SND_HDA_CODEC_SI3054 is not set CONFIG_SND_HDA_GENERIC=y CONFIG_SND_HDA_POWER_SAVE=y CONFIG_SND_HDA_POWER_SAVE_DEFAULT=120 # CONFIG_SND_HDSP is not set # CONFIG_SND_HDSPM is not set # CONFIG_SND_HIFIER is not set # CONFIG_SND_ICE1712 is not set # CONFIG_SND_ICE1724 is not set # CONFIG_SND_INTEL8X0 is not set # CONFIG_SND_INTEL8X0M is not set # CONFIG_SND_KORG1212 is not set # CONFIG_SND_MAESTRO3 is not set # CONFIG_SND_MIXART is not set # CONFIG_SND_NM256 is not set # CONFIG_SND_PCXHR is not set # CONFIG_SND_RIPTIDE is not set # CONFIG_SND_RME32 is not set # CONFIG_SND_RME96 is not set # CONFIG_SND_RME9652 is not set # CONFIG_SND_SONICVIBES is not set # CONFIG_SND_TRIDENT is not set # CONFIG_SND_VIA82XX is not set # CONFIG_SND_VIA82XX_MODEM is not set # CONFIG_SND_VIRTUOSO is not set # CONFIG_SND_VX222 is not set # CONFIG_SND_YMFPCI is not set # # USB devices # # CONFIG_SND_USB_AUDIO is not set # CONFIG_SND_USB_USX2Y is not set # CONFIG_SND_USB_CAIAQ is not set # # System on Chip audio support # # CONFIG_SND_SOC is not set # # SoC Audio support for SuperH # # # ALSA SoC audio for Freescale SOCs # # # Open Sound System # # CONFIG_SOUND_PRIME is not set # CONFIG_HID_SUPPORT is not set CONFIG_USB_SUPPORT=y CONFIG_USB_ARCH_HAS_HCD=y CONFIG_USB_ARCH_HAS_OHCI=y CONFIG_USB_ARCH_HAS_EHCI=y CONFIG_USB=m # CONFIG_USB_DEBUG is not set # CONFIG_USB_ANNOUNCE_NEW_DEVICES is not set # # Miscellaneous USB options # CONFIG_USB_DEVICEFS=y CONFIG_USB_DEVICE_CLASS=y # CONFIG_USB_DYNAMIC_MINORS is not set CONFIG_USB_SUSPEND=y CONFIG_USB_PERSIST=y # CONFIG_USB_OTG is not set # # USB Host Controller Drivers # CONFIG_USB_EHCI_HCD=m CONFIG_USB_EHCI_ROOT_HUB_TT=y CONFIG_USB_EHCI_TT_NEWSCHED=y # CONFIG_USB_ISP116X_HCD is not set CONFIG_USB_OHCI_HCD=m # CONFIG_USB_OHCI_BIG_ENDIAN_DESC is not set # CONFIG_USB_OHCI_BIG_ENDIAN_MMIO is not set CONFIG_USB_OHCI_LITTLE_ENDIAN=y # CONFIG_USB_UHCI_HCD is not set # CONFIG_USB_SL811_HCD is not set # CONFIG_USB_R8A66597_HCD is not set # # USB Device Class drivers # # CONFIG_USB_ACM is not set CONFIG_USB_PRINTER=m # # NOTE: USB_STORAGE enables SCSI, and 'SCSI disk support' # # # may also be needed; see USB_STORAGE Help for more information # CONFIG_USB_STORAGE=m # CONFIG_USB_STORAGE_DEBUG is not set # CONFIG_USB_STORAGE_DATAFAB is not set # CONFIG_USB_STORAGE_FREECOM is not set # CONFIG_USB_STORAGE_ISD200 is not set # CONFIG_USB_STORAGE_DPCM is not set # CONFIG_USB_STORAGE_USBAT is not set # CONFIG_USB_STORAGE_SDDR09 is not set # CONFIG_USB_STORAGE_SDDR55 is not set # CONFIG_USB_STORAGE_JUMPSHOT is not set # CONFIG_USB_STORAGE_ALAUDA is not set # CONFIG_USB_STORAGE_KARMA is not set CONFIG_USB_LIBUSUAL=y # # USB Imaging devices # # CONFIG_USB_MDC800 is not set # CONFIG_USB_MICROTEK is not set CONFIG_USB_MON=y # # USB port drivers # # CONFIG_USB_USS720 is not set # CONFIG_USB_SERIAL is not set # # USB Miscellaneous drivers # # CONFIG_USB_EMI62 is not set # CONFIG_USB_EMI26 is not set # CONFIG_USB_ADUTUX is not set # CONFIG_USB_AUERSWALD is not set # CONFIG_USB_RIO500 is not set # CONFIG_USB_LEGOTOWER is not set # CONFIG_USB_LCD is not set # CONFIG_USB_BERRY_CHARGE is not set # CONFIG_USB_LED is not set # CONFIG_USB_CYPRESS_CY7C63 is not set # CONFIG_USB_CYTHERM is not set # CONFIG_USB_PHIDGET is not set # CONFIG_USB_IDMOUSE is not set # CONFIG_USB_FTDI_ELAN is not set # CONFIG_USB_APPLEDISPLAY is not set # CONFIG_USB_SISUSBVGA is not set # CONFIG_USB_LD is not set # CONFIG_USB_TRANCEVIBRATOR is not set # CONFIG_USB_IOWARRIOR is not set # CONFIG_USB_TEST is not set # CONFIG_USB_GADGET is not set # CONFIG_MMC is not set # CONFIG_MEMSTICK is not set # CONFIG_NEW_LEDS is not set # CONFIG_INFINIBAND is not set # CONFIG_EDAC is not set # CONFIG_RTC_CLASS is not set # CONFIG_DMADEVICES is not set # CONFIG_AUXDISPLAY is not set # # Userspace I/O # # CONFIG_UIO is not set # # Firmware Drivers # # CONFIG_EDD is not set # CONFIG_DELL_RBU is not set # CONFIG_DCDBAS is not set CONFIG_DMIID=y # # File systems # CONFIG_EXT2_FS=y # CONFIG_EXT2_FS_XATTR is not set # CONFIG_EXT2_FS_XIP is not set CONFIG_EXT3_FS=y # CONFIG_EXT3_FS_XATTR is not set # CONFIG_EXT4DEV_FS is not set CONFIG_JBD=y # CONFIG_JBD_DEBUG is not set # CONFIG_REISERFS_FS is not set # CONFIG_JFS_FS is not set CONFIG_FS_POSIX_ACL=y # CONFIG_XFS_FS is not set # CONFIG_GFS2_FS is not set # CONFIG_OCFS2_FS is not set CONFIG_DNOTIFY=y CONFIG_INOTIFY=y CONFIG_INOTIFY_USER=y # CONFIG_QUOTA is not set # CONFIG_AUTOFS_FS is not set CONFIG_AUTOFS4_FS=y CONFIG_FUSE_FS=m # # CD-ROM/DVD Filesystems # CONFIG_ISO9660_FS=y CONFIG_JOLIET=y CONFIG_ZISOFS=y CONFIG_UDF_FS=y CONFIG_UDF_NLS=y # # DOS/FAT/NT Filesystems # CONFIG_FAT_FS=y CONFIG_MSDOS_FS=y CONFIG_VFAT_FS=y CONFIG_FAT_DEFAULT_CODEPAGE=437 CONFIG_FAT_DEFAULT_IOCHARSET="iso8859-1" # CONFIG_NTFS_FS is not set # # Pseudo filesystems # CONFIG_PROC_FS=y CONFIG_PROC_KCORE=y CONFIG_PROC_SYSCTL=y CONFIG_SYSFS=y CONFIG_TMPFS=y # CONFIG_TMPFS_POSIX_ACL is not set CONFIG_HUGETLBFS=y CONFIG_HUGETLB_PAGE=y # CONFIG_CONFIGFS_FS is not set # # Miscellaneous filesystems # # CONFIG_ADFS_FS is not set # CONFIG_AFFS_FS is not set # CONFIG_HFS_FS is not set # CONFIG_HFSPLUS_FS is not set # CONFIG_BEFS_FS is not set # CONFIG_BFS_FS is not set # CONFIG_EFS_FS is not set # CONFIG_CRAMFS is not set # CONFIG_VXFS_FS is not set # CONFIG_MINIX_FS is not set # CONFIG_HPFS_FS is not set # CONFIG_QNX4FS_FS is not set # CONFIG_ROMFS_FS is not set # CONFIG_SYSV_FS is not set # CONFIG_UFS_FS is not set CONFIG_NETWORK_FILESYSTEMS=y CONFIG_NFS_FS=y CONFIG_NFS_V3=y # CONFIG_NFS_V3_ACL is not set CONFIG_NFS_V4=y CONFIG_NFS_DIRECTIO=y CONFIG_NFSD=y CONFIG_NFSD_V3=y # CONFIG_NFSD_V3_ACL is not set CONFIG_NFSD_V4=y CONFIG_NFSD_TCP=y CONFIG_LOCKD=y CONFIG_LOCKD_V4=y CONFIG_EXPORTFS=y CONFIG_NFS_COMMON=y CONFIG_SUNRPC=y CONFIG_SUNRPC_GSS=y CONFIG_SUNRPC_BIND34=y CONFIG_RPCSEC_GSS_KRB5=y # CONFIG_RPCSEC_GSS_SPKM3 is not set # CONFIG_SMB_FS is not set CONFIG_CIFS=y # CONFIG_CIFS_STATS is not set # CONFIG_CIFS_WEAK_PW_HASH is not set CONFIG_CIFS_XATTR=y CONFIG_CIFS_POSIX=y # CONFIG_CIFS_DEBUG2 is not set CONFIG_CIFS_EXPERIMENTAL=y # CONFIG_NCP_FS is not set # CONFIG_CODA_FS is not set # CONFIG_AFS_FS is not set CONFIG_9P_FS=m # # Partition Types # # CONFIG_PARTITION_ADVANCED is not set CONFIG_MSDOS_PARTITION=y CONFIG_NLS=y CONFIG_NLS_DEFAULT="utf8" CONFIG_NLS_CODEPAGE_437=y # CONFIG_NLS_CODEPAGE_737 is not set # CONFIG_NLS_CODEPAGE_775 is not set CONFIG_NLS_CODEPAGE_850=y # CONFIG_NLS_CODEPAGE_852 is not set # CONFIG_NLS_CODEPAGE_855 is not set # CONFIG_NLS_CODEPAGE_857 is not set # CONFIG_NLS_CODEPAGE_860 is not set # CONFIG_NLS_CODEPAGE_861 is not set # CONFIG_NLS_CODEPAGE_862 is not set # CONFIG_NLS_CODEPAGE_863 is not set # CONFIG_NLS_CODEPAGE_864 is not set # CONFIG_NLS_CODEPAGE_865 is not set # CONFIG_NLS_CODEPAGE_866 is not set # CONFIG_NLS_CODEPAGE_869 is not set # CONFIG_NLS_CODEPAGE_936 is not set # CONFIG_NLS_CODEPAGE_950 is not set # CONFIG_NLS_CODEPAGE_932 is not set # CONFIG_NLS_CODEPAGE_949 is not set # CONFIG_NLS_CODEPAGE_874 is not set # CONFIG_NLS_ISO8859_8 is not set # CONFIG_NLS_CODEPAGE_1250 is not set # CONFIG_NLS_CODEPAGE_1251 is not set CONFIG_NLS_ASCII=y CONFIG_NLS_ISO8859_1=y # CONFIG_NLS_ISO8859_2 is not set # CONFIG_NLS_ISO8859_3 is not set # CONFIG_NLS_ISO8859_4 is not set # CONFIG_NLS_ISO8859_5 is not set # CONFIG_NLS_ISO8859_6 is not set # CONFIG_NLS_ISO8859_7 is not set # CONFIG_NLS_ISO8859_9 is not set # CONFIG_NLS_ISO8859_13 is not set # CONFIG_NLS_ISO8859_14 is not set CONFIG_NLS_ISO8859_15=y # CONFIG_NLS_KOI8_R is not set # CONFIG_NLS_KOI8_U is not set CONFIG_NLS_UTF8=y # CONFIG_DLM is not set # # Kernel hacking # CONFIG_TRACE_IRQFLAGS_SUPPORT=y # CONFIG_PRINTK_TIME is not set CONFIG_ENABLE_WARN_DEPRECATED=y # CONFIG_ENABLE_MUST_CHECK is not set CONFIG_MAGIC_SYSRQ=y # CONFIG_UNUSED_SYMBOLS is not set CONFIG_DEBUG_FS=y # CONFIG_HEADERS_CHECK is not set # CONFIG_DEBUG_KERNEL is not set # CONFIG_SLUB_DEBUG_ON is not set # CONFIG_SLUB_STATS is not set CONFIG_DEBUG_BUGVERBOSE=y # CONFIG_LATENCYTOP is not set # CONFIG_PROVIDE_OHCI1394_DMA_INIT is not set # CONFIG_SAMPLES is not set CONFIG_EARLY_PRINTK=y CONFIG_X86_MPPARSE=y CONFIG_IO_DELAY_TYPE_0X80=0 CONFIG_IO_DELAY_TYPE_0XED=1 CONFIG_IO_DELAY_TYPE_UDELAY=2 CONFIG_IO_DELAY_TYPE_NONE=3 CONFIG_IO_DELAY_0X80=y # CONFIG_IO_DELAY_0XED is not set # CONFIG_IO_DELAY_UDELAY is not set # CONFIG_IO_DELAY_NONE is not set CONFIG_DEFAULT_IO_DELAY_TYPE=0 # # Security options # # CONFIG_KEYS is not set # CONFIG_SECURITY is not set # CONFIG_SECURITY_FILE_CAPABILITIES is not set CONFIG_XOR_BLOCKS=y CONFIG_ASYNC_CORE=y CONFIG_ASYNC_MEMCPY=y CONFIG_ASYNC_XOR=y CONFIG_CRYPTO=y CONFIG_CRYPTO_ALGAPI=y CONFIG_CRYPTO_BLKCIPHER=y # CONFIG_CRYPTO_SEQIV is not set CONFIG_CRYPTO_MANAGER=y # CONFIG_CRYPTO_HMAC is not set # CONFIG_CRYPTO_XCBC is not set # CONFIG_CRYPTO_NULL is not set # CONFIG_CRYPTO_MD4 is not set CONFIG_CRYPTO_MD5=y # CONFIG_CRYPTO_SHA1 is not set # CONFIG_CRYPTO_SHA256 is not set # CONFIG_CRYPTO_SHA512 is not set # CONFIG_CRYPTO_WP512 is not set # CONFIG_CRYPTO_TGR192 is not set # CONFIG_CRYPTO_GF128MUL is not set # CONFIG_CRYPTO_ECB is not set CONFIG_CRYPTO_CBC=y # CONFIG_CRYPTO_PCBC is not set # CONFIG_CRYPTO_LRW is not set # CONFIG_CRYPTO_XTS is not set # CONFIG_CRYPTO_CTR is not set # CONFIG_CRYPTO_GCM is not set # CONFIG_CRYPTO_CCM is not set # CONFIG_CRYPTO_CRYPTD is not set CONFIG_CRYPTO_DES=y # CONFIG_CRYPTO_FCRYPT is not set # CONFIG_CRYPTO_BLOWFISH is not set # CONFIG_CRYPTO_TWOFISH is not set # CONFIG_CRYPTO_TWOFISH_X86_64 is not set # CONFIG_CRYPTO_SERPENT is not set # CONFIG_CRYPTO_AES is not set # CONFIG_CRYPTO_AES_X86_64 is not set # CONFIG_CRYPTO_CAST5 is not set # CONFIG_CRYPTO_CAST6 is not set # CONFIG_CRYPTO_TEA is not set # CONFIG_CRYPTO_ARC4 is not set # CONFIG_CRYPTO_KHAZAD is not set # CONFIG_CRYPTO_ANUBIS is not set # CONFIG_CRYPTO_SEED is not set # CONFIG_CRYPTO_SALSA20 is not set # CONFIG_CRYPTO_SALSA20_X86_64 is not set # CONFIG_CRYPTO_DEFLATE is not set # CONFIG_CRYPTO_MICHAEL_MIC is not set # CONFIG_CRYPTO_CRC32C is not set # CONFIG_CRYPTO_CAMELLIA is not set # CONFIG_CRYPTO_TEST is not set # CONFIG_CRYPTO_AUTHENC is not set # CONFIG_CRYPTO_LZO is not set # CONFIG_CRYPTO_HW is not set CONFIG_HAVE_KVM=y CONFIG_VIRTUALIZATION=y CONFIG_KVM=m # CONFIG_KVM_INTEL is not set CONFIG_KVM_AMD=m # CONFIG_VIRTIO_PCI is not set # CONFIG_VIRTIO_BALLOON is not set # # Library routines # CONFIG_BITREVERSE=y # CONFIG_CRC_CCITT is not set # CONFIG_CRC16 is not set # CONFIG_CRC_ITU_T is not set CONFIG_CRC32=y # CONFIG_CRC7 is not set # CONFIG_LIBCRC32C is not set CONFIG_ZLIB_INFLATE=y CONFIG_PLIST=y CONFIG_HAS_IOMEM=y CONFIG_HAS_IOPORT=y CONFIG_HAS_DMA=y -- (°= =°) //\ Prakash Punnoor /\\ V_/ \_V

2 1

[alsa-devel] [PATCH] hda: 92hd71bxxx DMIC nid
by Matthew Ranostay 20 Mar '08

20 Mar '08

Added missing DMIC verb to dell_4_1_pin_configs[]. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 6ffc943c3b39 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Tue Mar 18 17:18:18 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Tue Mar 18 14:09:17 2008 -0400 @@ -1317,7 +1317,7 @@ static unsigned int ref92hd71bxx_pin_con static unsigned int dell_m4_1_pin_configs[13] = { 0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110, - 0x23a1902e, 0x23014250, 0x40f000f0, 0x4f0000f0, + 0x23a1902e, 0x23014250, 0x40f000f0, 0x90a000f0, 0x40f000f0, 0x4f0000f0, };

2 1

[alsa-devel] [PATCH]model for alc262 to support Lenovo 3000
by Jiang zhe 20 Mar '08

20 Mar '08

Hi, Please refer to [0003856] on the alsa mantis. This model is to support the Lenovo 3000 y410. Thanks! Signed-off-by: Jiang zhe <zhe.jiang(a)intel.com>

2 1

Re: [alsa-devel] Random inverted audio channels
by Alexandre BOUIN 20 Mar '08

20 Mar '08

> I'm currently working on an internal project. > Here is some details : > This embedded project includes a AT91SAM9260 CPU and a WM8900 codec. > Audio driver is using ASoC API. > Audio data could be either switch from I2S or DSP A, but I prefer the > second one. > Audio Codec is master. > > We have an issue : when we play a right channel stereo track, sound is > played on right channel, but sometimes on left. > This issue could happen right after starting kernel, or later after 10- > 20 sounds played. Hi list, I'm back on this issue. I have few questions about it and I'd hope someone could answer : - does someone have already used AT91 SSC controller in slave mode ? Can he send me his register configuration and tell if he already heard and L/R channels swaps ? - is there a way to play a sound at the lowest software level, maybe without using alsa or dsp devices ? - where is the software audio buffer ? As you can see I really want to trace any difficulties to find where comes this problem. As always, if you have any other ideas to test, I'll be glad to hear them. Thanks and regards, Alex.

1 0

[alsa-devel] ALSA and SALSA install
by Alan Horstmann 19 Mar '08

19 Mar '08

Following on from thread 'Problems with ALSA in embedded device', running code such as aplay with both ALSA and SALSA libs. Is there a way to install both at the same time, so on a PC code can be compiled and experimented with first -lasound and then -lsalsa without alternately re-installing the libraries with make install? Thanks Alan

3 5

[alsa-devel] Same pci id for different ASUS machine
by Jiang zhe 19 Mar '08

19 Mar '08

Hi, According to the [0003839] https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3839 The PCI ID of ASUS F9S is 1043:1339 and "model=lenovo" works for it. But I found that ASUS G1 have the same id and model is 3stack in the alsa driver. Which one is true? --- alsa-driver-hg20080313/alsa-kernel/pci/hda/patch_realtek.c 2008-03-13 09:00:08.000000000 +0800 +++ b/alsa-kernel/pci/hda/patch_realtek.c 2008-03-13 10:38:57.000000000 +0800 @@ -12591,7 +12591,7 @@ static struct snd_pci_quirk alc861vd_cfg SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST), SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP), SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST), - SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST), + SND_PCI_QUIRK(0x1043, 0x1339, "Asus F9S", ALC861VD_LENOVO), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG), SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST), SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO),

4 5

[alsa-devel] HW decoding abstracted in ALSA
by Harsha priya gupta 19 Mar '08

19 Mar '08

Hi, I am new to ALSA and audio stack in Linux. I have a sound card that can do HW decoding as well. I am looking at writing an ALSA driver such that whatever may be the player or MMF above, it should use the HW decoder and not the inherit SW decoder that is available with the player. Can someone help me with some pointers or existing like-drivers? Thanks in advance, Harsha

2 2

[alsa-devel] About question of "ALSA lib pcm.c:2143:(snd_pcm_open_noupdate) Unknown PCM default"
by wangxf 19 Mar '08

19 Mar '08

alsa-devel，你好： I am new to ALSA, now I try to add one sound soc chip in linux 2.6.21 kernel on samsung arm designed board. I have do some underside work: 1: I have porting linux 2.6.21 kernel on samsung arm designed board.(log.txt add accessories) 2. kernel have recognise sound chip form log information. 3. I have a test app working in linux 2.6.18.(this test app is tested ok!) 4. I have download "alsa-lib-1.0.15.tar.bz2" and buid in linux host(Linux 2.6.18-5-686 #1 SMP Fri Jun 1 00:47:00 UTC 2007),create libasound.so.2.0.0. build order: 1)./configure --host=arm-linux CC=arm-linux-gcc ( i do not affirm this config) 2) make 3) and then change to su to use to make install. 5. download libasound.so.2.0.0 lib to samsung designed board and download test app to samsung designed board. 6. but i run test ,system prompt err: ALSA lib pcm.c:2143:(snd_pcm_open_noupdate) Unknown PCM default Playback open error: No such file or directory Segmentation fault now,I doubt libasound.so.2.0.0 lib is conflict with kernel 2.6.21. hope help . 夏新移动通讯有限公司软件设计工程师王小锋电话：86－592－6516777EXT6643 E-MAIL:wangxf@amoi.com.cn wangxf 2008-03-19

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[alsa-devel] Having a fight with snd_sctl_install:src/control/setup.c in alsa-lib-1.0.14a
by Bouterse, Bob 19 Mar '08

19 Mar '08

I am using alsa-lib-1.0.14a on a VIA EPIA 15000g board. I am hoping that someone can help me. I have been in an entrenched fight with the sctl functions setting my ctl setting back to the config file values. I am working on some extensions to the alsa audio output plugins in xine to provide simultaneous analog and iec958 on a Via EPIA 15000g, board. The issue is that I need to set the iec958 status bits at runtime, but everytime I call: /*---------------------------------------------------------------------- -----*/ ... snd_ctl_elem_value_get_iec958(cval, &iec958); llprintf(LOG_MIXER,"Read AES0=0x%02x,AES1=0x%02x,AES2=0x%02x,AES3=0x%02x\n", iec958.status[0], iec958.status[1], iec958.status[2], iec958.status[3]); /* AES0=0x6,AES1=0x82,AES2=0x0,AES3=0x2 */ iec958.status[0] = IEC958_AES0_NONAUDIO | IEC958_AES0_CON_NOT_COPYRIGHT; iec958.status[1] = IEC958_AES1_CON_PCM_CODER | IEC958_AES1_CON_ORIGINAL; iec958.status[2] = 0x00; iec958.status[3] = IEC958_AES3_CON_FS_48000; llprintf(LOG_MIXER,"Writing AES0=0x%02x,AES1=0x%02x,AES2=0x%02x,AES3=0x%02x\n", iec958.status[0], iec958.status[1], iec958.status[2], iec958.status[3]); /* store the values */ snd_ctl_elem_value_set_iec958(cval, &iec958); if ((err = snd_ctl_elem_write(ctl_handle, cval)) < 0) { xprintf (this->class->xine, XINE_VERBOSITY_DEBUG, "snd_ctl_elem_write(): %s\n", snd_strerror(err)); } ... /*---------------------------------------------------------------------- -----*/ snd_sctl_install resets the IEC958 status bits back to the original values! I am pulling my hair out trying to understand how to stop this. Unfortunately I don't have much time to fix this, please help. ---------------------------------------------------- Bob Bouterse Sr. Software Engineer Office of the CTO D&M Holdings Inc. 1-317-616-6541 (fax) 1-317-616-6541

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[alsa-devel] [PATCH] Use base ALC883 mixer for 6stack-dell model
by Herton Ronaldo Krzesinski 18 Mar '08

18 Mar '08

After DAC assignment fix in ALC883, alc888_6st_dell_mixer is now the same as alc883_base_mixer. Avoid duplicated code and use alc883_base_mixer in 6stack-dell model, removing alc888_6st_dell_mixer definition. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com> diff -p -up alsa-kernel/pci/hda/patch_realtek.c.orig alsa-kernel/pci/hda/patch_realtek.c --- alsa-kernel/pci/hda/patch_realtek.c.orig 2008-03-18 17:54:42.000000000 -0300 +++ alsa-kernel/pci/hda/patch_realtek.c 2008-03-18 18:00:08.000000000 -0300 @@ -6992,46 +6992,6 @@ static struct snd_kcontrol_new alc888_3s { } /* end */ }; -static struct snd_kcontrol_new alc888_6st_dell_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, 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_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = alc883_mux_enum_info, - .get = alc883_mux_enum_get, - .put = alc883_mux_enum_put, - }, - { } /* end */ -}; - static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -7941,7 +7901,7 @@ static struct alc_config_preset alc883_p .input_mux = &alc883_capture_source, }, [ALC888_6ST_DELL] = { - .mixers = { alc888_6st_dell_mixer, alc883_chmode_mixer }, + .mixers = { alc883_base_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc888_6st_dell_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, -- []'s Herton

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[alsa-devel] [PATCH] Remove now uneeded 6stack-hp model from ALC883
by Herton Ronaldo Krzesinski 18 Mar '08

18 Mar '08

After DAC assignment fix in ALC883, the 6stack-hp model is now the same as 6stack-dig. So just remove 6stack-hp model and replace its use with 6stack-dig. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com> diff -p -up alsa-kernel/Documentation/ALSA-Configuration.txt.orig alsa-kernel/Documentation/ALSA-Configuration.txt --- alsa-kernel/Documentation/ALSA-Configuration.txt.orig 2008-03-18 17:41:33.000000000 -0300 +++ alsa-kernel/Documentation/ALSA-Configuration.txt 2008-03-18 17:43:06.000000000 -0300 @@ -879,7 +879,6 @@ Prior to version 0.9.0rc4 options had a lenovo-nb0763 Lenovo NB0763 lenovo-ms7195-dig Lenovo MS7195 haier-w66 Haier W66 - 6stack-hp HP machines with 6stack (Nettle boards) 3stack-hp HP machines with 3stack (Lucknow, Samba boards) 6stack-dell Dell machines with 6stack (Inspiron 530) mitac Mitac 8252D diff -p -up alsa-kernel/pci/hda/patch_realtek.c.orig alsa-kernel/pci/hda/patch_realtek.c --- alsa-kernel/pci/hda/patch_realtek.c.orig 2008-03-18 17:30:24.000000000 -0300 +++ alsa-kernel/pci/hda/patch_realtek.c 2008-03-18 17:41:13.000000000 -0300 @@ -197,7 +197,6 @@ enum { ALC883_LENOVO_NB0763, ALC888_LENOVO_MS7195_DIG, ALC883_HAIER_W66, - ALC888_6ST_HP, ALC888_3ST_HP, ALC888_6ST_DELL, ALC883_MITAC, @@ -6955,46 +6954,6 @@ static struct snd_kcontrol_new alc883_me { } /* end */ }; -static struct snd_kcontrol_new alc888_6st_hp_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, 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_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = alc883_mux_enum_info, - .get = alc883_mux_enum_get, - .put = alc883_mux_enum_put, - }, - { } /* end */ -}; - static struct snd_kcontrol_new alc888_3st_hp_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -7720,7 +7679,6 @@ static const char *alc883_models[ALC883_ [ALC883_LENOVO_NB0763] = "lenovo-nb0763", [ALC888_LENOVO_MS7195_DIG] = "lenovo-ms7195-dig", [ALC883_HAIER_W66] = "haier-w66", - [ALC888_6ST_HP] = "6stack-hp", [ALC888_3ST_HP] = "3stack-hp", [ALC888_6ST_DELL] = "6stack-dell", [ALC883_MITAC] = "mitac", @@ -7739,7 +7697,7 @@ static struct snd_pci_quirk alc883_cfg_t SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG), SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP), - SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC888_6ST_HP), + SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC), @@ -7972,17 +7930,6 @@ static struct alc_config_preset alc883_p .unsol_event = alc883_haier_w66_unsol_event, .init_hook = alc883_haier_w66_automute, }, - [ALC888_6ST_HP] = { - .mixers = { alc888_6st_hp_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs }, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .dig_out_nid = ALC883_DIGOUT_NID, - .dig_in_nid = ALC883_DIGIN_NID, - .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), - .channel_mode = alc883_sixstack_modes, - .input_mux = &alc883_capture_source, - }, [ALC888_3ST_HP] = { .mixers = { alc888_3st_hp_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc888_3st_hp_verbs }, -- []'s Herton

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[alsa-devel] Issues with alsa-lib-1.0.13 built using arm gcc 4.1.1
by sandeep lahane 18 Mar '08

18 Mar '08

Hello, I have written an external io type plugin for an arm based board. While debugging the plugin, I encountered unexpected errors which I doubt are because of gcc optimizations, and after looking up on web it seems that arm-gcc 4.1.1 which we are using has some known bugs due to which alsa-library functions behave unexpectedly. E.g. snd_mask_refine() reportedly behaves differently when it is compiled using different optimization levels, I am facing the same issue where snd_mask_refine() returns unexpected (-EINVAL) and after I modify the code as in src/pcm/mask_inline.h MASK_INLINE void snd_mask_copy(snd_mask_t *mask, const snd_mask_t *v) { - *mask = *v; + memcpy(mask->bits, v->bits, MASK_SIZE * 4); } snd_mask_refine() returns 0 which is valid but I get segfaults just after returning from snd_pcm_ioplug_hw_refine(). Is this a known bug and if yes/no then is there any list of known bugs/quirks when using alsa-lib with arm gcc 4.1.1? I am attaching a log below, pardon for non-alsa flavour of debug messages in the log. # aplay -D myplug etc/startup.wav AlsaVersion=1.0.13, Opened for Playback ----- INFO (ALSA) : Successfully Registered the IO plugin ALSA-OPENMAX PLUGIN Playing WAVE 'etc/startup.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Ste reo ####aplay.c set_params calling snd_pcm_hw_params_any ###### snd_pcm_hw_params_any 1 ##### _snd_pcm_hw_params_any 0 ##### _snd_pcm_hw_params_any 2 ##### _snd_pcm_hw_params_any 8 ##### _snd_pcm_hw_params_any 19 _snd_pcm_hw_param_any in hw_is_mask _snd_pcm_hw_param_any in hw_is_mask _snd_pcm_hw_param_any in hw_is_mask _snd_pcm_hw_param_any in hw_is_interval _snd_pcm_hw_param_any in hw_is_interval _snd_pcm_hw_param_any in hw_is_interval _snd_pcm_hw_param_any in hw_is_interval _snd_pcm_hw_param_any in hw_is_interval _snd_pcm_hw_param_any in hw_is_interval _snd_pcm_hw_param_any in hw_is_interval _snd_pcm_hw_param_any in hw_is_interval _snd_pcm_hw_param_any in hw_is_interval _snd_pcm_hw_param_any in hw_is_interval _snd_pcm_hw_param_any in hw_is_interval _snd_pcm_hw_param_any in hw_is_interval ##### snd_pcm_hw_params_any 1 #####snd_pcm_hw_refine #####snd_pcm_hw_refine calling hw_refine #### snd_pcm_ioplug_hw_refine 1 snd_ext_parm_mask_refine 1 snd_mask_refine 1 snd_ext_parm_mask_refine return val is = 1 snd_ext_parm_mask_refine 1 snd_mask_refine 1 snd_ext_parm_mask_refine return val is = 1 snd_pcm_ioplug_hw_refine 2 snd_ext_parm_interval_refine 1 snd_ext_parm_interval_refine 1 snd_pcm_ioplug_hw_refine 3 snd_mask_refine 1 snd_mask_refine 1 snd_mask_refine 1 snd_pcm_ioplug_hw_refine 4 snd_ext_parm_interval_refine 1 snd_pcm_ioplug_hw_refine 5 snd_ext_parm_interval_refine 1 snd_ext_parm_interval_refine 1 snd_ext_parm_interval_refine 1 snd_ext_parm_interval_refine 1 snd_pcm_ioplug_hw_refine 5 change is = 1 ##### return val of hw_refine = 1 #####snd_pcm_hw_refine #####snd_pcm_hw_refine calling hw_refine #### snd_pcm_ioplug_hw_refine 1 snd_ext_parm_mask_refine 1 snd_mask_refine 1 snd_ext_parm_mask_refine return val is = 0 snd_ext_parm_mask_refine 1 snd_mask_refine 1 snd_ext_parm_mask_refine return val is = 0 snd_pcm_ioplug_hw_refine 2 snd_ext_parm_interval_refine 1 snd_ext_parm_interval_refine 1 snd_pcm_ioplug_hw_refine 3 snd_mask_refine 1 snd_pcm_ioplug_hw_refine 4 snd_ext_parm_interval_refine 1 snd_pcm_ioplug_hw_refine 5 snd_ext_parm_interval_refine 1 snd_ext_parm_interval_refine 1 snd_pcm_ioplug_hw_refine 5 change is = 0 ##### return val of hw_refine = 0 Dec 31 16:27:59 kernel: aplay: unhandled page fault (11) at 0x00000022, code 0x0 17 Dec 31 16:27:59 kernel: pgd = c30d8000 Dec 31 16:27:59 kernel: Dec 31 16:27:59 kernel: Dec 31 16:27:59 kernel: Pid: 206, comm: aplay Dec 31 16:27:59 kernel: CPU: 0 Not tainted (2.6.23.14-BROADCOM #1) Dec 31 16:27:59 kernel: PC is at 0x40170770 Dec 31 16:27:59 kernel: LR is at 0x40168210 Dec 31 16:27:59 kernel: pc : [<40170770>] lr : [<40168210>] psr: a0000010 Dec 31 16:27:59 kernel: sp : bee6f698 ip : 401855f8 fp : bee6fb64 Dec 31 16:27:59 kernel: r10: 401852b0 r9 : 00000006 r8 : bee6f894 Dec 31 16:27:59 kernel: r7 : 0001cc64 r6 : 00000022 r5 : 000130fc r4 : bee6f7 04 Dec 31 16:27:59 kernel: r3 : ffffffff r2 : 00000022 r1 : fffffffe r0 : 000000 22 Dec 31 16:27:59 kernel: Flags: NzCv IRQs on FIQs on Mode USER_32 ISA ARM Se gment user Dec 31 16:27:59 kernel: Control: 00e5387f Table: c30d8008 DAC: 00000015 Dec 31 16:27:59 kernel: (__do_user_fault+0x5c/0xa4) Dec 31 16:27:59 kernel: Dec 31 16:27:59 kernel: (do_page_fault+0x200/0x238) Dec 31 16:27:59 kernel: Dec 31 16:27:59 kernel: (do_DataAbort+0x3c/0xa0) Dec 31 16:27:59 kernel: (ret_from_exception+0x0/0x10) Dec 31 16:27:59 kernel: Exception stack(0xc31a1fb0 to 0xc31a1ff8) Dec 31 16:27:59 kernel: Dec 31 16:27:59 kernel: Dec 31 16:27:59 kernel: Dec 31 16:27:59 kernel: Segmentation fault Regards, Sandeep.

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[alsa-devel] If you have a copy callback, do you need to have a silence?
by Barclay, Bruce 18 Mar '08

18 Mar '08

I was finally able to get my driver working. I had to use the copy callback since my memory is external to Alsa. I have noticed that I commented out the silence callback and the ack. All seems to work and just for grins, I put them in with some printk's and they were never invoked. I don't fully understand these callbacks. When are they called and am I just lucky? Bruce --------------------- Bruce Barclay Intel Corporation ( Phone : 480-554-0553 ( Cell : 480-274-7493 * Email:bruce.barclay@intel.com * MS : CH7-308 "Real Gold Fears No Fire" - Li Quan

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[alsa-devel] arecord | aplay expected delay
by van der Linden 18 Mar '08

18 Mar '08

Hi, I'm running arecord and aplay under uCLinux on a Blackfin DSP. In this way I loopback my audio. Does anyone know what delay I can expect between audio input and audio output? I measure at my test board 50ms. I want to have about 1ms! E-J

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[alsa-devel] [PATCH] 1/2 alsa-python: API coverage documentation tool
by Aldrin Martoq 18 Mar '08

18 Mar '08

HG Log: ------- Added python tool of alsa-lib C API coverage for alsa-python binding. Description: ------------ The following patch is a python tool that shows how much of the asoundlib API is being covered by the alsa-python binding. It works by parsing the C source code and comparing with the official Doxygen documentation from alsa website. It will help the following users: - alsa-python developers: know how is mapped the original C API, so which python function/variable may be used. - python binding developers: know how much of the C API is covered. It could help for mistakes (free() instead of using snd_ctl_card_info_free), know what is missing, and statistic interest ;) Sample output: -------------- Functions --------- 1 const char* snd_asoundlib_version ( void ) Returns the ALSA sound library version in ASCII format. => Used in : alsacard.c: asoundlib_version Defines ------- N/A #define SND_CONTROL_DLSYM_VERSION _dlsym_control_001 dlsym version for interface entry callback **** NOT AVAILABLE/USED SND_CONTROL_DLSYM_VERSION **** 2 #define SND_CTL_ASYNC Async notification (flag for open mode) => As constant: alsacontrol.c: open_mode["ASYNC"] => As constant: alsahcontrol.c: open_mode["ASYNC"] 1 #define snd_ctl_card_info_alloca ( ptr ) allocate an invalid snd_ctl_card_info_t using standard alloca => Used in : alsacontrol.c: pyalsacontrol_cardinfo Stats: ------ After every section, the program shows some stats about usage. This is the current usage of each doxygen module: Global defines and functions 16 missing ( 94%) 0 excluded ( 0%) of 17 total ( 6% covered). Error handling 9 missing ( 90%) 0 excluded ( 0%) of 10 total ( 10% covered). Control Interface 137 missing ( 53%) 0 excluded ( 0%) of 257 total ( 47% covered). High level Control Interface 21 missing ( 46%) 0 excluded ( 0%) of 46 total ( 54% covered). Setup Control Interface 4 missing (100%) 0 excluded ( 0%) of 4 total ( 0% covered). MIDI Sequencer 25 missing ( 61%) 7 excluded ( 17%) of 41 total ( 39% covered). Sequencer Client Interface 15 missing ( 35%) 14 excluded ( 33%) of 43 total ( 65% covered). Sequencer Port Interface 48 missing ( 74%) 6 excluded ( 9%) of 65 total ( 26% covered). Sequencer Port Subscription 3 missing ( 7%) 15 excluded ( 33%) of 46 total ( 93% covered). Sequencer Queue Interface 58 missing ( 84%) 0 excluded ( 0%) of 69 total ( 16% covered). Sequencer Event API 42 missing ( 93%) 0 excluded ( 0%) of 45 total ( 7% covered). Sequencer Miscellaneous 0 missing ( N/A) 0 excluded ( N/A) of 0 total ( N/A covered). Sequencer Event Type Checks 7 missing ( 23%) 0 excluded ( 0%) of 30 total ( 77% covered). Sequencer Event Definitions 35 missing ( 40%) 0 excluded ( 0%) of 87 total ( 60% covered). Sequencer Middle Level Interfa 38 missing ( 78%) 0 excluded ( 0%) of 49 total ( 22% covered). Sequencer event <-> MIDI byte 11 missing (100%) 0 excluded ( 0%) of 11 total ( 0% covered). Mixer Interface 44 missing ( 75%) 0 excluded ( 0%) of 59 total ( 25% covered). Simple Mixer Interface 14 missing ( 18%) 0 excluded ( 0%) of 78 total ( 82% covered). Cheers! -- Aldrin Martoq <amartoq(a)dcc.uchile.cl>

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[alsa-devel] Query regarding SND_PCM_PLUGIN_DEFINE_FUNC()
by Pharaoh . 18 Mar '08

18 Mar '08

Hi, What does the mode field in SND_PCM_PLUGIN_DEFINE_FUNC() used for? is it for determining blocking mode or non-blocking mode of opening? -pharaoh

2 1

[alsa-devel] [PATCH] Removed deprecated sound/driver.h from Freescale MPC8610 drivers
by Timur Tabi 18 Mar '08

18 Mar '08

With commit 9004acc70e8c49c50c4c7b652f906f1e0ed5709d, include/sound/driver.h is deprecated. This patch removes the #include from fsl_ssi.c and fsl_dma.c. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/fsl_dma.c | 1 - sound/soc/fsl/fsl_ssi.c | 1 - 2 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 652514f..78de716 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -20,7 +20,6 @@ #include <linux/interrupt.h> #include <linux/delay.h> -#include <sound/driver.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 145ad13..b2a11b0 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -15,7 +15,6 @@ #include <linux/device.h> #include <linux/delay.h> -#include <sound/driver.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> -- 1.5.4

2 1

[alsa-devel] [PATCH] usbmidi.c: Call usb_autopm functions from rawmidi open/close callbacks
by Karsten Wiese 18 Mar '08

18 Mar '08

Lets "usb auto suspend" work for devices that set their usb_driver::supports_autosuspend to 1. Has no effect for other devices. Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de> --- sound/usb/usbmidi.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 6c616c7..842486f 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -860,11 +860,14 @@ static int snd_usbmidi_output_open(struct snd_rawmidi_substream *substream) } substream->runtime->private_data = port; port->state = STATE_UNKNOWN; + usb_autopm_get_interface(umidi->iface); return 0; } static int snd_usbmidi_output_close(struct snd_rawmidi_substream *substream) { + struct snd_usb_midi* umidi = substream->rmidi->private_data; + usb_autopm_put_interface(umidi->iface); return 0; } -- 1.5.3.3 -------------------------------------------------------

2 1

[alsa-devel] /usr/share/alsa/alsa.conf :and file extention
by Alexander Indenbaum 18 Mar '08

18 Mar '08

Hello, I'm playing with embedded system without any hardware sound device. I'd like to grab sound from ALSA application, for instance Firefox displaying YouTube clip inside flash plugin and to store output PCM stream into file on local disk. I got an advice to use File extension. So I defined following /usr/share/alsa/alsa.conf pcm.default{ type file file /tmp/pcm.out } But when alsa lib still complains about "Unknown PCM plughw:0,0" "Cannot open shared library /usr/lib/alsa-lib/libasound_module_pcm_file.so" I have a nasty feeling that syntax of my /usr/share/alsa/alsa.conf is broken beyond recognition. I wonder what is the correct way to define my situation: * No real audio devices * All sound is output goes to file as PCM stream Do you have a clue? Alexander Indenbaum

5 17

[alsa-devel] ~ALSA Audio coordinator/manager
by Harsha priya gupta 18 Mar '08

18 Mar '08

Hi all, I am looking to see if anyone is working on an Audio manager kind of solution as a part of ALSA. 'By Audio manager, I mean a coordinator of all audio input and output streams. A coordinator which is application and devices aware . Example functionality of an audio manager would to direct different steams to different devices like (voice output to headset and music output to speakers). Or to pause an media player when a voice recording starts.' If anyone is doing any work on these lines, please let me know about the work and its integration into ALSA as well. Thanks in Advance, Harsha.

2 1

[alsa-devel] basic questions
by Harsha priya gupta 18 Mar '08

18 Mar '08

Hi, I am ramping up on ALSA architecture and framework. I have a couple of basic questions. I apologize if these questions are very basic to be raised in this mailing list. If anyone can help me find answers, it would be great. Thanks in advance. 1. My assumption is that ALSA handles only PCM streams today. Does it handle encoded streams from application to sound card (Which can do hardware decoding)? 2. What are the hwdep files in ALSA used for? Where do the concerned IOCTLS land to? Thanks and Regards, Harsha.

3 4

[alsa-devel] [PATCH] Add missing audio path between Mono Mixer and Mic PGAs.
by Mark Brown 18 Mar '08

18 Mar '08

From: Robert Jarzmik <rjarzmik(a)free.fr> Signed-off-by: Robert Jarzmik <rjarzmik(a)free.fr> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9713.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index e2949c5..172ca1a 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -500,6 +500,8 @@ static const char *audio_map[][3] = { {"Mono Mixer", "Aux Playback Switch", "Aux DAC"}, {"Mono Mixer", "Bypass Playback Switch", "Line Mixer"}, {"Mono Mixer", "PCM Playback Switch", "AC97 Mixer"}, + {"Mono Mixer", "Mic 1 Sidetone Switch", "Mic A PGA"}, + {"Mono Mixer", "Mic 2 Sidetone Switch", "Mic B PGA"}, {"Mono Mixer", NULL, "Capture Mono Mux"}, /* DAC inv mux 1 */ -- 1.5.4.4

2 1

[alsa-devel] [PATCH] Map 3stack-6ch-dig ALC883 model for MSI 945GCM5 V2 (MSI-7267)
by Herton Ronaldo Krzesinski 18 Mar '08

18 Mar '08

Map 3stack-6ch-dig ALC883 model for MSI 945GCM5 V2 (MSI-7267). Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com> diff -p -up alsa-kernel/pci/hda/patch_realtek.c.orig alsa-kernel/pci/hda/patch_realtek.c --- alsa-kernel/pci/hda/patch_realtek.c.orig 2008-03-18 00:18:52.000000000 -0300 +++ alsa-kernel/pci/hda/patch_realtek.c 2008-03-18 00:48:18.000000000 -0300 @@ -7763,6 +7763,7 @@ static struct snd_pci_quirk alc883_cfg_t SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7250, "MSI", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1462, 0x7267, "MSI", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG), -- []'s Herton

2 1

[alsa-devel] [PATCH] Fix DAC assignment order in ALC883
by Herton Ronaldo Krzesinski 18 Mar '08

18 Mar '08

Actually clfe and surround DACs are inverted in alc883_dac_nids array (see ALC883 datasheet). I discovered this while testing multi-channel setup (using 3stack-6ch-dig model) on MSI 945GCM5 V2 motherboard that has an ALC883 codec. Simply Rear Left/Right and Center/LFE were swapped in 6 channel mode (also in 4 channel mode you didn't get rear left/right output). Other models also were affected by this bug, as can be seen by the mixer layouts that "workaround" this (the real bug was not noticed, and some other models simply played with mixer and initial verbs). Thus along with fixing the order of dac nids, also change the models that relied on previous dac ordering properly. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com> diff -p -up alsa-kernel/pci/hda/patch_realtek.c.orig alsa-kernel/pci/hda/patch_realtek.c --- alsa-kernel/pci/hda/patch_realtek.c.orig 2008-03-18 00:16:12.000000000 -0300 +++ alsa-kernel/pci/hda/patch_realtek.c 2008-03-18 00:18:52.000000000 -0300 @@ -6450,7 +6450,7 @@ static int patch_alc882(struct hda_codec static hda_nid_t alc883_dac_nids[4] = { /* front, rear, clfe, rear_surr */ - 0x02, 0x04, 0x03, 0x05 + 0x02, 0x03, 0x04, 0x05 }; static hda_nid_t alc883_adc_nids[2] = { @@ -6958,12 +6958,12 @@ static struct snd_kcontrol_new alc883_me static struct snd_kcontrol_new alc888_6st_hp_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), @@ -6998,12 +6998,12 @@ static struct snd_kcontrol_new alc888_6s static struct snd_kcontrol_new alc888_3st_hp_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), @@ -7036,12 +7036,12 @@ static struct snd_kcontrol_new alc888_3s static struct snd_kcontrol_new alc888_6st_dell_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), @@ -7332,26 +7332,14 @@ static struct hda_verb alc883_haier_w66_ { } /* end */ }; -static struct hda_verb alc888_6st_hp_verbs[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Rear : output 2 (0x0e) */ - {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* CLFE : output 1 (0x0d) */ - {0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, /* Side : output 3 (0x0f) */ - { } -}; - static struct hda_verb alc888_3st_hp_verbs[] = { {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ - {0x18, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */ - {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CLFE : output 2 (0x0e) */ + {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */ + {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CLFE : output 2 (0x0e) */ { } }; static struct hda_verb alc888_6st_dell_verbs[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Rear : output 1 (0x0e) */ - {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* CLFE : output 2 (0x0d) */ - {0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, /* Side : output 3 (0x0f) */ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, { } }; @@ -7985,7 +7973,7 @@ static struct alc_config_preset alc883_p }, [ALC888_6ST_HP] = { .mixers = { alc888_6st_hp_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs, alc888_6st_hp_verbs }, + .init_verbs = { alc883_init_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, -- []'s Herton

2 1

[alsa-devel] query about snd_pcm_hw_refine
by Pharaoh . 18 Mar '08

18 Mar '08

Hi I get following error when trying to run aplay with my plugin: #aplay -D myplug test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, St ereo aplay: set_params:882: Broken configuration for this PCM: no configurations available. After looking at the code, I figured out that its snd_pcm_hw_refine which is causing problem. It looks like following in pcm_params.c in alsa lib 1.0.13, int snd_pcm_hw_refine(snd_pcm_t *pcm, snd_pcm_hw_params_t *params) { int res; #ifdef REFINE_DEBUG snd_output_t *log; snd_output_stdio_attach(&log, stderr, 0); #endif printf("#####%s \n", __FUNCTION__); assert(pcm && params); #ifdef REFINE_DEBUG snd_output_printf(log, "REFINE called:\n"); snd_pcm_hw_params_dump(params, log); #endif printf("#####%s calling hw_refine\n", __FUNCTION__); res = pcm->ops->hw_refine(pcm->op_arg, params); printf("#####% return val of hw_refine = %d\n", res); #ifdef REFINE_DEBUG snd_output_printf(log, "refine done - result = %i\n", res); snd_pcm_hw_params_dump(params, log); snd_output_close(log); #endif return res; } I have modified the alsa.conf for my plugin by adding following in it at the end, pcm.myplug { type myplug card 0 } What could be the problem? -pharaoh.

1 3

[alsa-devel] DbAu1200/1210/1250 AC97 Driver
by Chris Gray 17 Mar '08

17 Mar '08

Hello, After looking for DbAu12xx AC97 ALSA support I ported the existing drivers to support Au1200 AC'97 PSC configuration. Being new to the development mailing list, is there a protocol on how to post patches for this sort of thing? Is this even the right forum? Thanks in advance! -Chris

1 0

[alsa-devel] no front speaker sound with ALC262
by Rene Dohmen 17 Mar '08

17 Mar '08

Dear List, I have a problem with an Intel HDA soundcard lspci output: ----- 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 01) The sound volume is very very low when i use the headphone output. The device has integrated speakers but i don't hear any sound coming out of them (maybe because the volume is so low...). I've put all the mixer channels to the maximum values. When booted in Win XP the soundcard works correctly. Is see one line in my logs that indicate that something is wrong: hda_codec: Unknown model for ALC262, trying auto-probe from BIOS... I didn't have it working with other versions of ALSA. Tested with 1.0.14(provided with ubunty gutsy), then started reading, lots of suggestions stated that the problem would be solved in 1.0.15, so i compiled 1.0.16 from src and replaced the default alsa modules with the new ones. De device i 'm using is supplied by tangent ( http://www.tangent.com/t_all_in_one_LCDPC/vita_8000s.htm) its an all in one solution with touchscreen and speakers integrated in one small cabinet. I runned the alsa-info.sh script output and uploaded it: http://pastebin.ca/913865 I don't know if this is the correct place for getting help but all help is appreciated, Best Regards, Rene Dohmen

5 14

[alsa-devel] Emvy24control s/pidf input issue
by Udo van den Heuvel 17 Mar '08

17 Mar '08

Hello, I just noticed that S/PDIF input on my Audiophile 2496 does not work. I see: [udo@wda ~]$ envy24control using --- input_channels: 0 --- output_channels: 4 --- pcm_output_channels: 8 --- spdif in/out channels: 2 Unable to read S/PDIF input switch: No such file or directory Changing the Digital Input from coax to optical or vice versa gives: Unable to write S/PDIF on/off switch: No such file or directory Unable to write S/PDIF input switch: No such file or directory This is on Fedora 8 x86_64 with alsa-tools 1.0.12-4 from Fedora Core 7, envycontrol 0.6.0. S/PDIF output works without a problem. Did not have S/PDIF input working before, as far as I remember. Using -v I do not see signal in the meters. In the source I see in spdif_input_update: if ((card_eeprom.subvendor != ICE1712_SUBDEVICE_DELTADIO2496) && ! card_is_dmx6fire) return; if ((err = snd_ctl_elem_read(ctl, spdif_input)) < 0) g_print("Unable to read S/PDIF input switch: %s\n", snd_strerror(err)); (etc) Why does snd_ctl_elem_read fail? How could I make S/PDIF input work again? Tips are very welcome! Kind regards, Udo

2 4

[alsa-devel] AD1988B: ALSA doesn't use dmix automatically
by Dâniel Fraga 17 Mar '08

17 Mar '08

ALSA 1.0.15 http://alsa.opensrc.org/index.php/Dmix NOTE: For ALSA 1.0.9rc2 and higher you don't need to setup dmix. Dmix is enabled as default for soundcards which don't support hw mixing. So why if I try to play 2 sounds at the same time, the second one complains the "resource is busy"? Shouldn't it detect that AD1988B can't do hardware mixing and use dmix automatically? Obs.: I don't have any ~/.asoundrc or /etc/asound.conf lying around. Thanks! -- Linux 2.6.24: Arr Matey! A Hairy Bilge Rat! http://u-br.net http://www.abusar.org/FELIZ_2008.html Napalm Death - "Circle Of Hypocrisy" (Harmony Corruption - 1990)

4 4

Re: [alsa-devel] ALSACTL wrong restore on ARM9
by Liam Girdwood 17 Mar '08

17 Mar '08

On Fri, 2008-03-07 at 16:28 +0100, Alexandre BOUIN wrote: > > Can you describe 'bad values' in a little more detail. > > You're right, some details are missing. > > For example, for first control, a playback volume control, value can go from > 0 to 63. > Before "store" command, I'd set it to 20. > I've check that in "/var/lib/alsa/asound/state", I had the same value. Here > is a part of this file : > > state.codec { > control.1 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 63' > iface MIXER > name 'HP Playback Volume' > value.0 20 > value.1 20 > } > > So saved value seems ok. > > After "restore" command, I get this : > > $amixer cget numid=1 > numid=1,iface=MIXER,name='HP Playback Volume' > ; type=INTEGER,access=rw------,values=2,min=0,max=63,step=0 > : values=45,45 > > That's why I say that values are not correct. > I'm currently using debian alsa-utils_1.0.16-1_arm package. > It's possible that your codec drivers register caching is borked. This can be easily confirmed with debug in your driver kcontrol code. If your driver looks ok, then I'd suggest looking into your alsa-lib cross build. There have been some problems with it being incorrectly cross compiled for ARM in the past (pls search the list for details). Fwiw, I use OpenEmbedded to build ALSA for ARM and I've never ran into this issue. Liam

2 2

[alsa-devel] PATCH - some fixes and cleanup for ICE1724 cards
by Pavel Hofman 17 Mar '08

17 Mar '08

Hi Takashi, here is a small patch with minor fixes and cleanup. * removing the hack with NON_AKM ak4xxx type * support for card-specific flags in ak4114_stats * definition of the flags for corresponding cards Signed-by: Pavel Hofman <dustin(a)seznam.cz> Thanks, Pavel

2 1

[alsa-devel] [PATCH] [ML403-AC97CR] Remove duplicate snd_card_set_dev().
by Joachim Foerster 17 Mar '08

17 Mar '08

Hi Takashi, I noticed, that there was a change in ml403-ac97cr.c, which introduces a second call of snd_card_set_dev() in the _probe()-function. The "original" one is in the snd_ml403_ac97cr_create()-function. According the other changes, I guess, we want to have it in the _probe()-function, now (=> following patch)? From: Joachim Foerster <JOFT(a)gmx.de> We want to have snd_card_set_dev() in _probe(), but not a second one in snd_ml403_ac97cr_create(). Signed-off-by: Joachim Foerster <JOFT(a)gmx.de> --- sound/drivers/ml403-ac97cr.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c index 05a871a..b654007 100644 --- a/sound/drivers/ml403-ac97cr.c +++ b/sound/drivers/ml403-ac97cr.c @@ -1191,8 +1191,6 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev, return err; } - snd_card_set_dev(card, &pfdev->dev); - *rml403_ac97cr = ml403_ac97cr; return 0; } -- 1.5.4.3

2 1

[alsa-devel] Grab PCM stream into file
by Alexander Indenbaum 17 Mar '08

17 Mar '08

Hello, I'm playing with embedded system without any hardware sound device. Strategically we'd like to forward PCM stream from ALSA clients on user level to proprietary application. As initial proof of concept stage I'd like to grab sound from ALSA application, for instance Firefox displaying YouTube clip inside flash plugin and to store output PCM stream into file on local disk. Please advice how I could achieve this goal. Alexander Indenbaum

1 0

[alsa-devel] How to stop playback on alsa synth?
by Lukasz Pawelczyk 16 Mar '08

16 Mar '08

Hello. I'd like to ask how to completely stop playback using alsa synth. Basically my program initialized output in the following way: snd_seq_open() snd_seq_set_client_name() snd_seq_client_id() snd_seq_parse_address() snd_seq_create_simple_port() snd_seq_connect_to() snd_seq_alloc_named_queue() Then I create a new thread that processes events. It sets tempo and in infinite loop it does: snd_seq_ev_set_fixed() snd_seq_event_output() This all is based on aplaymidi code. How can I asynchronously immediately stop all the playback? Basicaly it requires breaking snd_seq_event_output() thats locked on my thread. I tried everything I could come up with. I know very little about alsa synth programming and basically thats the only thing i need to do now. Tried: snd_seq_ev_set_queue_stop() snd_seq_stop_queue() snd_seq_remove_events() or even closing the seq completely to reopen it on another file playback with: snd_seq_stop_queue() snd_seq_disconnect_to() snd_seq_delete_simple_port() snd_seq_close() All failed, results were very various. I simply run out of ideas. It doesn't matter if the method would require to close seq or not. I just need to stop the playback so i can gracefully stop the thread and play another file (with eventual reopening ports/seq). Would appreciate any help. -- Regards Havner {jid,mail}:havner(at)pld-linux.org "Quis custodiet ipsos custodes?"

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[PATCH] Correct dapm controls length.
by Robert Jarzmik 16 Mar '08

16 Mar '08

Dapm control names should follow alsa standard, which defines controls name's length to 44 and not 32. Signed-off-by: Robert Jarzmik <rjarzmik(a)free.fr> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 4b03e3f..a61024c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -281,7 +281,7 @@ static int dapm_new_mixer(struct snd_soc_card *soc_card, struct snd_soc_dapm_widget *w) { int i, ret = 0; - char name[32]; + char name[44]; struct snd_soc_dapm_path *path; /* add kcontrol */ @@ -295,7 +295,7 @@ static int dapm_new_mixer(struct snd_soc_card *soc_card, continue; /* add dapm control with long name */ - snprintf(name, 32, "%s %s", w->name, w->kcontrols[i].name); + snprintf(name, 44, "%s %s", w->name, w->kcontrols[i].name); path->long_name = kstrdup (name, GFP_KERNEL); if (path->long_name == NULL) return -ENOMEM; -- 1.5.6.5

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[PATCH] Dirty hack to use GPIO95 in AC97 reset.
by Robert Jarzmik 16 Mar '08

16 Mar '08

Mio Mitac A701 uses GPIO95 instead of GPIO113 as the reset line connected to the WM9713 chip. This is dirty, and should be replaced by an elegant solution where cold and warm reset are implemented in machine driver mioa701_wm9713.c Signed-off-by: Robert Jarzmik <rjarzmik(a)free.fr> --- sound/soc/pxa/pxa2xx-ac97.c | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index cb12544..86fb9c2 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -145,10 +145,10 @@ static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) #ifdef CONFIG_PXA27x /* warm reset broken on Bulverde, so manually keep AC97 reset high */ - pxa_gpio_mode(113 | GPIO_OUT | GPIO_DFLT_HIGH); + pxa_gpio_mode(95 | GPIO_OUT | GPIO_DFLT_HIGH); udelay(10); GCR |= GCR_WARM_RST; - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + pxa_gpio_mode(95 | GPIO_ALT_FN_1_OUT); udelay(500); #elif defined(CONFIG_PXA3xx) /* Can't use interrupts */ @@ -303,8 +303,8 @@ static int pxa2xx_ac97_resume(struct platform_device *dev) pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD); #endif #ifdef CONFIG_PXA27x - /* Use GPIO 113 as AC97 Reset on Bulverde */ - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + /* Use GPIO 95 as AC97 Reset on Bulverde */ + pxa_gpio_mode(95 | GPIO_ALT_FN_1_OUT); #endif clk_enable(ac97_clk); return 0; @@ -457,8 +457,8 @@ static int pxa2xx_ac97_probe(struct platform_device *pdev) return -ENOMEM; #ifdef CONFIG_PXA27x - /* Use GPIO 113 as AC97 Reset on Bulverde */ - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + /* Use GPIO 95 as AC97 Reset on Bulverde */ + pxa_gpio_mode(95 | GPIO_ALT_FN_1_OUT); ac97conf_clk = clk_get(&pdev->dev, "AC97CONFCLK"); if (IS_ERR(ac97conf_clk)) { -- 1.5.5.3 --=-=-= Content-Type: text/x-diff Content-Disposition: inline; filename=0002-Dirty-hack-to-workaround-pxa27x-buggy-hardreset.patch

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[alsa-devel] Maya44 driver in the official ALSA tree
by Piotr Zaryk 15 Mar '08

15 Mar '08

Hi, When will be the next alsa-driver version released? Can't wait till maya44 support will be added to the official alsa tree. It will help with the development of the new driver. -- Piotr Zaryk

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Re: [alsa-devel] no front speaker sound with ALC262
by Takashi Iwai 14 Mar '08

14 Mar '08

At Fri, 29 Feb 2008 17:24:03 +0100, Rene Dohmen wrote: > > No I still don't know where the speakers are conencted. > I asked the support engineers at tangent to send me a detailed > blockdiagram of their sound solution and ik think i'll receive that > sooner or later. > > So, for now, I just tried passing the EAPD option to all NIDS in the > codec#0 file. > > I attached the code#0 file while working with very low volume > > When I run > sudo hda-verb/hda-verb /dev/snd/hwC0D0 0x03 SET_EAPD 0x02 > or > sudo hda-verb/hda-verb /dev/snd/hwC0D0 0x02 SET_EAPD 0x02 Doh, I actually misunderstood that your codec were ALC268 but it's ALC262. EAPD command can be issued only on widgets that support. ALC262 has no EAPD pin caps at all. so it doesn't work like that. Instead, ALC262 (and other Realtek codecs) has a special sequence with coefficients to the vendor-defined widget. Yes, it's a black magic. Try the following sequence: 0x20 SET_COEF_INDEX 0x07 0x20 SET_PROC_COEF 0x3060 0x3060 can be different numbers like 0x3070 or 0x3050. If you set a different number, reset the coef index again (i.e. always call as a pair). Takashi

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[alsa-devel] aplay: playback:2021: read error
by Pharaoh . 14 Mar '08

14 Mar '08

Hi, I am getting aplay: playback:2021: read error while running my plugin on an arm based board, it seems that, this has to do something with permissions etc or may be the sound card is being shared etc. I don't have a registered sound card since there is no kernel driver for my sound hw. I use my plugin as, aplay -D my_plug test.wav -pharaoh.

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[alsa-devel] AK4114 - capturing spdif input stops the stream
by Pavel Hofman 14 Mar '08

14 Mar '08

Hi, here is my scenario: ICE1724 card, trying to record from SPDIF input via AK4114, ICE1724 slaved to SPDIF clock from the receiver. The card detects SPDIF input rate correctly (in my case ESI Juli where AK4114 is provided with independent clock signal to enable the rate detection). Let's say it is 192000. Now I want to arecord the input stream, let's say in CD quality, using the plug plugin. The command fails, because of the check in ak4114.c: res = external_rate(rcs1); if (!(flags & AK4114_CHECK_NO_RATE) && runtime && runtime->rate != res) { snd_pcm_stream_lock_irqsave(ak4114->capture_substream, _flags); if (snd_pcm_running(ak4114->capture_substream)) { printk(KERN_DEBUG "rate changed (%i <- %i)\n", runtime->rate, res); snd_pcm_stop(ak4114->capture_substream, SNDRV_PCM_STATE_DRAINING); res = 1; } snd_pcm_stream_unlock_irqrestore(ak4114->capture_substream, _flags); } The stream gets stopped because res = 192000 and runtime->rate = 44100. The problem is that the capture device still offers all the available rates, instead of the single SPDIF input one. Thus, the plug plugin is not forced to convert from 192000 to 44100 and runtime->rate is 44100 instead of 192000. Of course recording at 192000 works fine. What would be the best way to force the SPDIF capture device to offer (advertise) only the current rate? This functionality would have to be applicable only to the few cards correctly detecting incoming rate (unlike most ice1724 cards without independent clock in slave mode). If the card cannot detect input frequency reliably (the driver developer should know), I guess it should not do the check in ak4114.c at all - I can prepare a simple patch for that. The same will most probably apply to cards with AK4117 as the code looks the same. I know it is a minor issue but if a user hits this problem, it will take him a lot of effort to fix his setup appropriately. Thanks, Pavel.

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[alsa-devel] [ALC262 patch_realtek.c] Fujitsu Lifebook port replicator/dock headphone jack sense
by Tony Vroon 14 Mar '08

14 Mar '08

Good afternoon, Could you please consider the following patch for the 2.6.26 merge window. The docking station headphone output had no audio and jack sense was not considered. Jack information from the laptop itself and the dock are combined, as the dock does not obscure the connector. Signed-Off-By: Tony Vroon <tony(a)linx.net> Regards, Tony V. LINX sysadmin P.S. Codec graph included in case anyone wants to start using mic jack sense for automated capture source selection.

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[alsa-devel] Underrun errors in aplay
by Barclay, Bruce 14 Mar '08

14 Mar '08

I have written a driver for Alsa and I think the sound is correct but I do get underrun errors. I don't understand how aplay would know if I am starved. This doesn't make sense to me. I would only think that the driver should know about underruns but maybe I am confused with this. I am wondering if I am calling snd_pcm_period_elapsed too often. I am calling that when I have free H/W buffers. Thanks, Bruce

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[alsa-devel] reading peak meter values from a RME HDSP madi soundcard
by ListsAccount 14 Mar '08

14 Mar '08

Hi, I'd like to read the peak values of the RME HDSP madi soundcard. To do this, I use the function: snd_hwdep_ioctl() with the ioctl command "SNDRV_HDSPM_IOCTL_GET_PEAK_RMS". I use the following code: // create container hdspm_peak_rms_t madi_peak_rms; hdspm_peak_rms_t* ptr_madi_peak_rms = &madi_peak_rms; //snd_hwdep_open ... // fill the container with values snd_hwdep_ioctl(hw, SNDRV_HDSPM_IOCTL_GET_PEAK_RMS, (void *)&ptr_madi_peak_rms); for (unsigned int chn = 0; chn < numberOfChannels; ++chn) { //put values from madi soundcard into std::vector containers currentInputPeak[chn] = madi_peak_rms.level_offset[chn]; currentPlaybackPeak[chn] = madi_peak_rms.level_offset[chn+64]; currentOutputPeak[chn] = madi_peak_rms.level_offset[chn+128]; } The upper 24 bit of the read out values should be the signal peak. The next 8 bit seem to represent a clipping counter. For the first eight input channels this works pretty well. But for every eighth channel of all further channels (input, playback and output) the peak value is to low and is not in time with the other channels. Do i use the function in the right way? Is there a special function to get the peak meter values? (In a similar way as for ADAT soundcards?) Best regards.

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[alsa-devel] bugfix for RME HDSPM alsa driver
by ListsAccount 14 Mar '08

14 Mar '08

Hi, I have had the following problem with the RME HDSPM alsa driver: settings of card: Preferred Sync Reference = Word Sample Clock Source = AutoSync and no input plugged in (neither madi nor optical) playing a file (important!) Monitoring the external rate with amixer, the value was jumping between "None" and "48000 kHz". The proc output showed that the sample rate on word clock was still right at 48000 kHz while the external rate was jumping here, too. External rate seems to jump between word clock and madi input select. When "Preferred Sync Reference = Word" is selected, just the word clock rate should be displayed! I fixed the problem with following patch in alsa-driver-1.0.16/alsa-kernel/pci/rme9652/hdspm.c in function hdspm_external_sample_rate() line 668: There was a typo: instead of status2, status was used. 668c668 < (status & HDSPM_SelSyncRef0) == 0) { --- > (status2 & HDSPM_SelSyncRef0) == 0) { (same way as in snd_hdspm_proc_read_madi function line 3123) greetings

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