September 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Error patching file pcm_native.c
by Delusion's Master 20 Nov '24

20 Nov '24

Hi, first of all I apologize in case you receive multiple copies of this message, but I forgot to specify a subject in the original mail. I have a problem compiling the latest alsa driver snapshots. Specifically, it's a problem when patching pcm_native.c: [..] copying file alsa-kernel/core/info.c patching file info.c copying file alsa-kernel/core/pcm.c patching file pcm.c Hunk #3 succeeded at 739 (offset 2 lines). Hunk #5 succeeded at 954 (offset 2 lines). Hunk #7 succeeded at 1045 (offset 2 lines). copying file alsa-kernel/core/pcm_native.c patching file pcm_native.c Hunk #4 succeeded at 1544 (offset -25 lines). Hunk #5 succeeded at 2827 (offset 12 lines). Hunk #6 succeeded at 2878 (offset 10 lines). Hunk #7 FAILED at 2906. Hunk #8 FAILED at 2922. Hunk #9 succeeded at 2915 (offset -23 lines). Hunk #10 succeeded at 3036 (offset 10 lines). Hunk #11 succeeded at 3027 (offset -23 lines). Hunk #12 succeeded at 3074 (offset 10 lines). Hunk #13 succeeded at 3098 (offset -23 lines). Hunk #14 succeeded at 3143 (offset 10 lines). Hunk #15 succeeded at 3124 (offset -23 lines). Hunk #16 succeeded at 3214 (offset 10 lines). Hunk #17 succeeded at 3208 (offset -23 lines). Hunk #18 succeeded at 3271 (offset 10 lines). Hunk #19 succeeded at 3304 (offset -23 lines). Hunk #20 succeeded at 3386 (offset 10 lines). Hunk #21 succeeded at 3411 (offset -23 lines). Hunk #22 succeeded at 3456 (offset 10 lines). Hunk #23 succeeded at 3450 (offset -23 lines). Hunk #24 succeeded at 3525 (offset 10 lines). Hunk #25 succeeded at 3644 (offset -23 lines). 2 out of 25 hunks FAILED -- saving rejects to file pcm_native.c.rej make[2]: *** [pcm_native.c] Error 1 It's some days I get this, but I waited to report this since I thought it could be a temporary problem. My OS is Fedora 11 x86_64. Previous snapshots didn't give me this problem, but I can't say which one broke the compilation, since I usually try updated snapshots only at kernel updates. I searched for this error and the only relevant report I could find is this one: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-April/016420.html a report from last April. What could the problem be? Thanks _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx…

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[alsa-devel] hda-intel missing mixer channels
by Orm Finnendahl 26 Aug '13

26 Aug '13

Hi, I sent this to alsa-user a week ago but didn't get any reply. Does anybody know, what is responsible for the missing faders? Did the driver change or is it some module loading mechanism in the distro which prevents the faders from showing up in alsamixer? This is so strange. Please help! -- Orm -------------------------------------------------------------------- Hi everybody, after the last update of Ubuntu (intrepid to jaunty) on my Lenovo Ideapad my microphone input doesn't work anymore (neither internal nor external). I found out that alsamixer lacks quite a lot of faders in the newer kernel/module compared to the older one. Some of the faders now unavailable were crucial for Mic input to work on the older system so I'd very much like to get them back ;-) Some more infos: Soundcard: Intel 82801G (ICH7 Family) Chip: Realtek ALC269 Module: snd_hda_intel (in both kernels) Older kernel: Ubuntu stock 2.6.27-14-generic Newer kernel: Ubuntu stock 2.6.28-13-rt Is there any way to make the missing controls available or did the driver change and I have to compile the older driver for the new kernel? Yours, Orm

2 3

[alsa-devel] [ASoc] TDM over one SSC channel on AT91 possible?
by Friedrich Lobenstock 24 Jun '11

24 Jun '11

Hello! The background: I'm just designing an ATMEL AT91RM9200 based board which got two TLV320AIC33 on board. It is either possible to hook up the two TLV320AIC33 to a separate SSC (synchronous serial control) channel each or use one SSC channel and configure it for TDM (time division multiplex) mode. There is already a driver for TLV320AIC14k/TLV320AIC2k in the development tree, so adding support for the TLV320AIC33 should be feasible. What I'm not sure about is the fact if ASOC is prepared for using channels, especially SSC, in TDM mode. Is the core ASOC structure prepared to handle that? Would we be better off instantiate one drive twice each time with different parameters for the SSC channel? Thanks for any feedback. -- MfG / Regards Friedrich Lobenstock _________________________________________________________ SCOTTY Group Austria GmbH a SCOTTY Group plc company, http://www.scottygroup.com/ _________________________________________________________

4 3

[alsa-devel] idt STAC927x adat supported (sigmatel HDA codec) ?
by Eric 03 Mar '11

03 Mar '11

Is the ADAT output of the IDT (sigmatel) STAC927x HDA codec supported in alsa ? If not could someone please try to add this incredibly nice feature in the existing alsa driver ? Thank you very much

4 13

[alsa-devel] davinci-i2c,pcm updates
by Troy Kisky 27 Sep '10

27 Sep '10

This patch set is against git://git.alsa-project.org/alsa-kernel.git but the last patch in series won't compile until the sram allocator patch is available.

8 44

Re: [alsa-devel] Patchmix-style application for EMU cards
by Camilo Polymeris 30 Aug '10

30 Aug '10

Hello > That is more or less what I had figured out, but it how are playback_1 to > > _16 channels routed to DSP0-15? It seems that it depends on the setting > of > > Front, Surround, LFE, etc. mixer controls. If they are turned up, I > don't > > understand what happens, sound to playback channels appears on various > DSP > > buses, at different levels. > > Really? Well, I didn't much experiment with surround mode. In source code, > I > can see that there are up to playback 8 channels at hw:0,0 with can be > controlled by faders in alsamixer. I tried surround40 and surroun51 on my > emu1616m and it worked as expected. The channles were mapped to first > couple > of DSP ports in a correct order. In fact this should work the same as > audigy > does, except the result is passed to DSP ports instead of physical > outputs. > On my card, playback channels 1 & 2 (hw:0,3) appear on all DSP busses associated with surround. Say you turn up the "Side" fader in alsamixer, channels 1 & 2 will appear at the relevant DSPs (6-7, I think) Whatever, for now I am just keeping all surround controls down (that is, Front, Surround, Center, Side & LFE). > Anyway, I think if there is another layer of routing done in audigy DSP, > it > shouldn't be used in EMU specific application. Or at least not for now. > The > most important thing is to provide all the switches and FPGA routing > scheme > for each EMU model. Or, to be precise, three routing schemes for each EMU > model, because for "double speed" (e.g. 96kHz) and "quadro speed" (e.g. > 192kHz) modes of operation the numebr of both physical and virtual > channels > changes. > I think the switches and enums for the card models are rather trivial, one can handle it with a customized version of qamix (done with a simple XML file), for example. A separate application would not be necessary for only that. > > If I turn the Front/Surround/LFE/etc. faders down, I can mix playback > > channels to DSP using the "Multichannel PCM Send/Routing/Volume" faders, > in > > a more or less intuitive way, besides that routing values do not map > > directly to DSP buses (i.e. 14-15 map to DSP0-1, but 2-3 map to DSP2-3). > Handling of this controls motivated me to write the application in the first place. No sound at all will appear if I don't set the appropiate controls for each playback channel: Multichannel PCM Volume (full), Multichannel PCM Send (an index of 8 controls, the first 4 select the target DSP) & Multichannel PCM Send Volume (again faders 0-3 of 8) By the way, only DSP0-7 work in this way. I have no idea how to make DSP8-31 work. On which card you did tested it? The driver was primary written for the > 1820M/1212M V1, so some things may not work well on other models and need > to > be fixed. I have a 1212m, don't know which version, but I think it's V1-- it does have firewire. I am using driver version 1.0.16. Thanks, Camilo.

3 3

[alsa-devel] snd-hda-intel: no data when recording from mic, "IRQ timing workaround is activated for card #0. Suggest a bigger bdl_pos_adj" logged in kern.log
by Aleksander Adamowski 17 Mar '10

17 Mar '10

Hi! I think I have some interesting problems to analyse for Takashi. Maybe my sound card model is a non typical one, but I have a continuous stream of capture-related problems with it. It's an integrated sound card on my Gigabyte GA-MA74GM-S2H Rev. 1.1 motherboard. "grep ^Codec /proc/asound/card0/codec#0" shows: Codec: Realtek ALC888 I'm attaching the full contents of /proc/asound/card0/codec#0 for identification. But first, some background. With this particular card, I've been continuously hit by the "azx_get_response timeout, switching to single_cmd mode" problem, described here: http://www.kernel.org/pub/linux/kernel/people/tiwai/docs/HD-Audio.html#_cod… However, in my case setting probe_mask=1 (experimentally determined) didn't make the problem go away, although it seemed to reduce its frequency (might be just a coincidence). Whenever the dreaded "azx_get_response timeout, switching to single_cmd mode" message appeared in the kernel log, recording stopped working. The card got into the bad state where it would not record sound, and arecord would be giving a stream of "overrun" error messages: arecord -f cd output.wav Recording WAVE 'a.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo overrun!!! (at least 27.876 ms long) overrun!!! (at least 26.965 ms long) overrun!!! (at least 16.169 ms long) overrun!!! (at least 11.749 ms long) overrun!!! (at least 11.298 ms long) overrun!!! (at least 11.814 ms long) overrun!!! (at least 11.190 ms long) overrun!!! (at least 11.190 ms long) overrun!!! (at least 11.747 ms long) overrun!!! (at least 11.819 ms long) overrun!!! (at least 11.149 ms long) overrun!!! (at least 11.257 ms long) overrun!!! (at least 11.315 ms long) overrun!!! (at least 11.651 ms long) overrun!!! (at least 11.745 ms long) overrun!!! (at least 11.235 ms long) overrun!!! (at least 11.169 ms long) overrun!!! (at least 11.776 ms long) overrun!!! (at least 11.391 ms long) overrun!!! (at least 11.669 ms long) overrun!!! (at least 11.714 ms long) overrun!!! (at least 11.671 ms long) overrun!!! (at least 11.190 ms long) What's notable, sound playback would continue to work OK. After being fed up with interrupted Skype conversations (where I weren't immediately aware that the other side stopped hearing me, since I was hearing them), I've decided to upgrade do kernel 2.6.29 (citing the http://www.kernel.org/pub/linux/kernel/people/tiwai/docs/HD-Audio.html#_cod… document: "Since 2.6.29 kernel, the driver has a more robust probing method, so this error might happen rarely, though"). After upgrading to 2.6.29 I've removed any extra snd-hda-intel options from modprobe config files. Now the azx_get_response timeout seems to be gone, but I'm experiencing another problem: after some powered up time, recording starts giving empty streams - there's no error, but audio data has zero lengh. e.g. when I arecord, I get a .wav file that is only 44 bytes long (which seems to be the length of the WAV header). When I make a Skype test call to echo123, and it starts playing back my message, it's zero length too - the playback end signal can be heard immediately after the playback start signal. This problem seems to concide with the following error message in the kernel log: hda-intel: IRQ timing workaround is activated for card #0. Suggest a bigger bdl_pos_adj. Do you have any idea what could be the cause and how to debug this issue further? -- Best Regards, Aleksander Adamowski http://olo.org.pl

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[alsa-devel] [PATCH] ARM: DaVinci: Audio support for DA850/OMAP-L138 EVM
by Chaithrika U S 20 Jan '10

20 Jan '10

Define resources for McASP used on DA850/OMAP-L138 EVM, add platform device defintion and Pin Mux configurations. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies on DaVinci GIT tree. This patch is dependent on the following patch which I have submitted earlier: [PATCH v2] ARM: DaVinci: Audio support for DA830 EVM arch/arm/mach-davinci/board-da850-evm.c | 29 ++++++++++++++++++++++ arch/arm/mach-davinci/da850.c | 39 ++++++++++++++++++++++++++++++ arch/arm/mach-davinci/devices-da8xx.c | 34 +++++++++++++++++++++++++- arch/arm/mach-davinci/include/mach/asp.h | 7 +++++ arch/arm/mach-davinci/include/mach/mux.h | 25 +++++++++++++++++++ 5 files changed, 133 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-davinci/board-da850-evm.c b/arch/arm/mach-davinci/board-da850-evm.c index d989346..52bfe4c 100644 --- a/arch/arm/mach-davinci/board-da850-evm.c +++ b/arch/arm/mach-davinci/board-da850-evm.c @@ -38,6 +38,28 @@ static struct davinci_uart_config da850_evm_uart_config __initdata = { .enabled_uarts = 0x7, }; +/* davinci da850 evm audio machine driver */ +static u8 da850_iis_serializer_direction[] = { + INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, + INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, + INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, TX_MODE, + RX_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, +}; + +static struct snd_platform_data da850_evm_snd_data = { + .tx_dma_offset = 0x2000, + .rx_dma_offset = 0x2000, + .op_mode = DAVINCI_MCASP_IIS_MODE, + .num_serializer = ARRAY_SIZE(da850_iis_serializer_direction), + .tdm_slots = 2, + .serial_dir = da850_iis_serializer_direction, + .eventq_no = EVENTQ_1, + .version = MCASP_VERSION_2, + .txnumevt = 1, + .rxnumevt = 1, +}; + + static __init void da850_evm_init(void) { struct davinci_soc_info *soc_info = &davinci_soc_info; @@ -86,6 +108,13 @@ static __init void da850_evm_init(void) */ __raw_writel(0, IO_ADDRESS(DA8XX_UART1_BASE) + 0x30); __raw_writel(0, IO_ADDRESS(DA8XX_UART0_BASE) + 0x30); + + ret = da8xx_pinmux_setup(da850_mcasp_pins); + if (ret) + pr_warning("da850_evm_init: mcasp mux setup failed: %d\n", + ret); + + da8xx_init_mcasp(0, &da850_evm_snd_data); } #ifdef CONFIG_SERIAL_8250_CONSOLE diff --git a/arch/arm/mach-davinci/da850.c b/arch/arm/mach-davinci/da850.c index 4a43ae2..e33b226 100644 --- a/arch/arm/mach-davinci/da850.c +++ b/arch/arm/mach-davinci/da850.c @@ -289,6 +289,13 @@ static struct clk emac_clk = { .lpsc = DA8XX_LPSC1_CPGMAC, }; +static struct clk mcasp_clk = { + .name = "mcasp", + .parent = &pll0_sysclk2, + .lpsc = DA8XX_LPSC1_McASP0, + .psc_ctlr = 1, +}; + static struct davinci_clk da850_clks[] = { CLK(NULL, "ref", &ref_clk), CLK(NULL, "pll0", &pll0_clk), @@ -326,6 +333,7 @@ static struct davinci_clk da850_clks[] = { CLK(NULL, "arm", &arm_clk), CLK(NULL, "rmii", &rmii_clk), CLK("davinci_emac.1", NULL, &emac_clk), + CLK("davinci-mcasp.0", NULL, &mcasp_clk), CLK(NULL, NULL, NULL), }; @@ -370,6 +378,30 @@ static const struct mux_config da850_pins[] = { MUX_CFG(DA850, MII_RXD_2, 3, 20, 15, 8, false) MUX_CFG(DA850, MII_RXD_1, 3, 24, 15, 8, false) MUX_CFG(DA850, MII_RXD_0, 3, 28, 15, 8, false) + /* McASP function */ + MUX_CFG(DA850, ACLKR, 0, 0, 15, 1, false) + MUX_CFG(DA850, ACLKX, 0, 4, 15, 1, false) + MUX_CFG(DA850, AFSR, 0, 8, 15, 1, false) + MUX_CFG(DA850, AFSX, 0, 12, 15, 1, false) + MUX_CFG(DA850, AHCLKR, 0, 16, 15, 1, false) + MUX_CFG(DA850, AHCLKX, 0, 20, 15, 1, false) + MUX_CFG(DA850, AMUTE, 0, 24, 15, 1, false) + MUX_CFG(DA850, AXR_15, 1, 0, 15, 1, false) + MUX_CFG(DA850, AXR_14, 1, 4, 15, 1, false) + MUX_CFG(DA850, AXR_13, 1, 8, 15, 1, false) + MUX_CFG(DA850, AXR_12, 1, 12, 15, 1, false) + MUX_CFG(DA850, AXR_11, 1, 16, 15, 1, false) + MUX_CFG(DA850, AXR_10, 1, 20, 15, 1, false) + MUX_CFG(DA850, AXR_9, 1, 24, 15, 1, false) + MUX_CFG(DA850, AXR_8, 1, 28, 15, 1, false) + MUX_CFG(DA850, AXR_7, 2, 0, 15, 1, false) + MUX_CFG(DA850, AXR_6, 2, 4, 15, 1, false) + MUX_CFG(DA850, AXR_5, 2, 8, 15, 1, false) + MUX_CFG(DA850, AXR_4, 2, 12, 15, 1, false) + MUX_CFG(DA850, AXR_3, 2, 16, 15, 1, false) + MUX_CFG(DA850, AXR_2, 2, 20, 15, 1, false) + MUX_CFG(DA850, AXR_1, 2, 24, 15, 1, false) + MUX_CFG(DA850, AXR_0, 2, 28, 15, 1, false) #endif }; @@ -406,6 +438,13 @@ const short da850_cpgmac_pins[] __initdata = { -1 }; +const short da850_mcasp_pins[] __initdata = { + DA850_AHCLKX, DA850_ACLKX, DA850_AFSX, + DA850_AHCLKR, DA850_ACLKR, DA850_AFSR, DA850_AMUTE, + DA850_AXR_11, DA850_AXR_12, + -1 +}; + /* FIQ are pri 0-1; otherwise 2-7, with 7 lowest priority */ static u8 da850_default_priorities[DA850_N_CP_INTC_IRQ] = { [IRQ_DA8XX_COMMTX] = 7, diff --git a/arch/arm/mach-davinci/devices-da8xx.c b/arch/arm/mach-davinci/devices-da8xx.c index fe0baaf..1882eef 100644 --- a/arch/arm/mach-davinci/devices-da8xx.c +++ b/arch/arm/mach-davinci/devices-da8xx.c @@ -309,6 +309,34 @@ static struct platform_device da830_mcasp1_device = { .resource = da830_mcasp1_resources, }; +static struct resource da850_mcasp_resources[] = { + { + .name = "mcasp", + .start = DAVINCI_DA8XX_MCASP0_REG_BASE, + .end = DAVINCI_DA8XX_MCASP0_REG_BASE + (SZ_1K * 12) - 1, + .flags = IORESOURCE_MEM, + }, + /* TX event */ + { + .start = DAVINCI_DA8XX_DMA_MCASP0_AXEVT, + .end = DAVINCI_DA8XX_DMA_MCASP0_AXEVT, + .flags = IORESOURCE_DMA, + }, + /* RX event */ + { + .start = DAVINCI_DA8XX_DMA_MCASP0_AREVT, + .end = DAVINCI_DA8XX_DMA_MCASP0_AREVT, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device da850_mcasp_device = { + .name = "davinci-mcasp", + .id = 0, + .num_resources = ARRAY_SIZE(da850_mcasp_resources), + .resource = da850_mcasp_resources, +}; + int __init da8xx_register_emac(void) { return platform_device_register(&da8xx_emac_device); @@ -316,8 +344,12 @@ int __init da8xx_register_emac(void) void __init da8xx_init_mcasp(int id, struct snd_platform_data *pdata) { - if (id == 1) { /* DA830/OMAP-L137 has 3 instances of McASP */ + /* DA830/OMAP-L137 has 3 instances of McASP */ + if (cpu_is_davinci_da830() && id == 1) { da830_mcasp1_device.dev.platform_data = pdata; platform_device_register(&da830_mcasp1_device); + } else if (cpu_is_davinci_da850()) { + da850_mcasp_device.dev.platform_data = pdata; + platform_device_register(&da850_mcasp_device); } } diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index f3c97ac..18e4ce3 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -15,6 +15,9 @@ #define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000 #define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800 +/* Bases of da850/da830 McASP0 register banks */ +#define DAVINCI_DA8XX_MCASP0_REG_BASE 0x01D00000 + /* Bases of da830 McASP1 register banks */ #define DAVINCI_DA830_MCASP1_REG_BASE 0x01D04000 @@ -29,6 +32,10 @@ #define DAVINCI_DM646X_DMA_MCASP0_AREVT0 9 #define DAVINCI_DM646X_DMA_MCASP1_AXEVT1 12 +/* EDMA channels of da850/da830 McASP0 */ +#define DAVINCI_DA8XX_DMA_MCASP0_AREVT 0 +#define DAVINCI_DA8XX_DMA_MCASP0_AXEVT 1 + /* EDMA channels of da830 McASP1 */ #define DAVINCI_DA830_DMA_MCASP1_AREVT 2 #define DAVINCI_DA830_DMA_MCASP1_AXEVT 3 diff --git a/arch/arm/mach-davinci/include/mach/mux.h b/arch/arm/mach-davinci/include/mach/mux.h index a676b2f..f50f5b2 100644 --- a/arch/arm/mach-davinci/include/mach/mux.h +++ b/arch/arm/mach-davinci/include/mach/mux.h @@ -748,6 +748,31 @@ enum davinci_da850_index { DA850_MII_RXD_2, DA850_MII_RXD_1, DA850_MII_RXD_0, + + /* McASP function */ + DA850_ACLKR, + DA850_ACLKX, + DA850_AFSR, + DA850_AFSX, + DA850_AHCLKR, + DA850_AHCLKX, + DA850_AMUTE, + DA850_AXR_15, + DA850_AXR_14, + DA850_AXR_13, + DA850_AXR_12, + DA850_AXR_11, + DA850_AXR_10, + DA850_AXR_9, + DA850_AXR_8, + DA850_AXR_7, + DA850_AXR_6, + DA850_AXR_5, + DA850_AXR_4, + DA850_AXR_3, + DA850_AXR_2, + DA850_AXR_1, + DA850_AXR_0, }; #ifdef CONFIG_DAVINCI_MUX -- 1.5.6

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[alsa-devel] Audio buffer mmap-ing on sh arch vs. cache aliasing problem
by Pawel MOLL 16 Jan '10

16 Jan '10

Hi Guys, Recently I was investigating an issue with capturing audio using USB Audio Class device on a sh4-based board. "Bad voice quality" was reported... Finally I have traced the problem to something which is (unfortunately) well known to sh developers as a D-cache aliasing (or synonym) problem. Briefly speaking: due to some MMU design decisions, one can have two different virtual address pointing to the same physical location, which is fine, but going via different cache slots! So if there was a value of "0" in the memory and user "A" will write "1" there, user "B" will still read "0"... The solution is to ensure all TLB entries (so virtual memory areas) are beginning from a 16kB-aligned virtual address. Otherwise it is necessary to flush the cache between accesses from "A" and "B" sides. And now. The USB Audio Class driver (sound/usb/usbaudio.c) is allocating the sound buffer like this... static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t size) { [...] runtime->dma_area = vmalloc(size); [...] } ... and vmalloc will return a page(4k)-aligned pointer, possibly not 16k-aligned one. This is the source of all evil ;-) When using RW transfers everything is fine, as data is memcopied between two independent buffers. In a case of MMAP mode we will end up with such a situation: 1. The driver will memcpy data from an URB to the buffer. It will populate several cache lines. Let's call it a "kernel cache". 2. As the library has mapped the non-16k-aligned address via different cache line ("user cache"), it will read some rubbish from the physical memory, populating the "user cache" by the way. 3. Some time later the "kernel cache" is flushed, writing the data to the memory. 4. Some new data from URB is entering "kernel cache". 5. The library will access mmap-ed area again, going via the "user cache", which may or may not reflect the correct data (depending on the fact was the "user cache" flushed or not) etc. Of course this cycle is completely not deterministic, so some of the "kernel cache" lines will be flushed before being accessed from user space, other not... The final effect is... hmm... bizarre :-) First, of course, you will get a (probably loud) glitch (rubbish from buffer's underlying memory, before the first valid data is written back), and then something that could be described as an "echo" ;-) I mean - you are capturing "1 2 3 4 5 6 7..." and the result is (G stands for Glitch ;-) "G 1 23 3 4 4 6 67..." As a quick-and-dirty work-around I have modified snd_pcm_alloc_vmalloc_buffer() to allocate always 12kB more and then use 16k-aligned pointer: static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t size) { struct snd_pcm_runtime *runtime = subs->runtime; if (runtime->dma_addr) { if (runtime->dma_bytes >= size) return 0; /* already large enough */ vfree((void *)runtime->dma_addr); } runtime->dma_addr = (unsigned long)vmalloc(size + 12 * 1024); if (!runtime->dma_addr) return -ENOMEM; runtime->dma_area = (void *)ALIGN(runtime->dma_addr, 16 * 1024); runtime->dma_bytes = size; return 0; } Of course it cannot be regarded as anything more as a hack. And definitely not as a solution... So my question is: Any idea how the "proper solution" should look like? I see no obvious method to get an arch-independent "mmap-compliant" buffer. This problem was found on a sh arch, using an USB Audio Class device, however out there some other architectures are suffering from this MMU "feature" as well (ie. see http://docs.hp.com/en/B3906-90006/ch07s09.html) and possibly other drivers could behave "wrongly" (quoted as the driver is actually innocent...) To be honest I just have absolutely no idea what to do with this all! :-O I hope I was clear enough in the description... Any feedback, advise, idea etc. will be more than appreciated. Cheers Paweł

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[alsa-devel] Intel G45 8-Channel LPCM over HDMI
by Alexander Werner 11 Dec '09

11 Dec '09

Hi, I'm planning to build a linux-based, small sized PC for hearing aid acousticians. It would be highly unfavorable, if the connection between Receiver and PC was analog, because of the lower quality and the high number of cables. I searched for a appropriate mainboard, and found the DG45FC, which fits my needs perfectly, apart from the missing audio over HDMI support. Is there any plan to support multichannel LPCM output over HDMI on the Intel G45 platform? If not, is there anyone willing to implement HDMI support for the G45 chipset if I donate the needed hardware (permanently)? But if there's no chance to implement it in the near future, has anyone experience with an onboard chipset with multichannel LPCM over HDMI and ALSA support? Thanks, Alexander Werner

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[alsa-devel] ALSA on Alienware M17x
by Adam Gray 06 Dec '09

06 Dec '09

Hi guys, I'm having a lot of trouble getting sound out of my Alienware M17x. I've attached the output from alsa-info. I'm running kernel version 2.6.28 with ALSA driver version 1.0.18. There's no sound at all from the speakers or the headphones. In alsamixer, all channels appear un-muted and turned up full. If you need any further information please let me know. Regards, Adam Gray

3 19

[alsa-devel] [PATCH 2/4] ASoC: DaVinci: Voice Codec Interface
by miguel.aguilar＠ridgerun.com 25 Nov '09

25 Nov '09

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> 1) Adds an interface needed by the voice codec CQ0093. 2) Add an option to select internal or external codec in the DM365. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- sound/soc/davinci/Kconfig | 26 +++- sound/soc/davinci/Makefile | 2 + sound/soc/davinci/davinci-vcif.c | 363 ++++++++++++++++++++++++++++++++++++++ sound/soc/davinci/davinci-vcif.h | 63 +++++++ 4 files changed, 452 insertions(+), 2 deletions(-) create mode 100644 sound/soc/davinci/davinci-vcif.c create mode 100644 sound/soc/davinci/davinci-vcif.h diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 047ee39..051a10e 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -9,18 +9,40 @@ config SND_DAVINCI_SOC config SND_DAVINCI_SOC_I2S tristate +config SND_DAVINCI_SOC_VCIF + tristate + config SND_DAVINCI_SOC_MCASP tristate config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" depends on SND_DAVINCI_SOC - depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM + depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM select SND_DAVINCI_SOC_I2S select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on TI - DaVinci DM6446 or DM355 EVM platforms. + DaVinci DM6446, DM355 or DM365 EVM platforms. + +choice + prompt "DM365 codec select" + depends on SND_DAVINCI_SOC_EVM + depends on MACH_DAVINCI_DM365_EVM + default SND_DM365_EXTERNAL_CODEC + +config SND_DM365_INTERNAL_CODEC + bool "CQ93VC" + select SND_DAVINCI_SOC_VCIF + select SND_SOC_CQ0093VC + help + Say Y if you want to add support for SoC On-chip voice codec + +config SND_DM365_EXTERNAL_CODEC + bool "AIC3101" + help + Say Y if you want to add support for AIC3101 audio codec +endchoice config SND_DM6467_SOC_EVM tristate "SoC Audio support for DaVinci DM6467 EVM" diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index a6939d7..a93679d 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -2,10 +2,12 @@ snd-soc-davinci-objs := davinci-pcm.o snd-soc-davinci-i2s-objs := davinci-i2s.o snd-soc-davinci-mcasp-objs:= davinci-mcasp.o +snd-soc-davinci-vcif-objs:= davinci-vcif.o obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o +obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o # DAVINCI Machine Support snd-soc-evm-objs := davinci-evm.o diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c new file mode 100644 index 0000000..83191f1 --- /dev/null +++ b/sound/soc/davinci/davinci-vcif.c @@ -0,0 +1,363 @@ +/* + * ALSA SoC Voice Codec Interface for TI DAVINCI processor + * + * Copyright (C) 2009 Texas Instruments. + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * Initial code: Hui Geng + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "davinci-pcm.h" +#include "davinci-i2s.h" +#include "davinci-vcif.h" + +#define MOD_REG_BIT(val, mask, set) do { \ + if (set) { \ + val |= mask; \ + } else { \ + val &= ~mask; \ + } \ +} while (0) + +static struct davinci_pcm_dma_params davinci_vcif_pcm_out = { + .name = "VCIF PCM Stereo out", +}; + +static struct davinci_pcm_dma_params davinci_vcif_pcm_in = { + .name = "VCIF PCM Stereo in", +}; + +struct davinci_vcif_dev { + void __iomem *base; + resource_size_t pbase; + size_t base_size; + int mode; + u32 pcr; + struct clk *clk; + struct davinci_pcm_dma_params *dma_params[2]; +}; + +static inline void davinci_vcif_write_reg(struct davinci_vcif_dev *dev, + int reg, u32 val) +{ + __raw_writel(val, dev->base + reg); +} + +static inline u32 davinci_vcif_read_reg(struct davinci_vcif_dev *dev, int reg) +{ + return __raw_readl(dev->base + reg); +} + +static void davinci_vcif_start(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_vcif_dev *dev = rtd->dai->cpu_dai->private_data; + u32 w; + + /* Start the sample generator and enable transmitter/receiver */ + w = davinci_vcif_read_reg(dev, DAVINCI_VCIF_CTRL_REG); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + MOD_REG_BIT(w, DAVINCI_VCIF_CTRL_RSTDAC, 1); + else + MOD_REG_BIT(w, DAVINCI_VCIF_CTRL_RSTADC, 1); + + davinci_vcif_write_reg(dev, DAVINCI_VCIF_CTRL_REG, w); +} + +static void davinci_vcif_stop(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_vcif_dev *dev = rtd->dai->cpu_dai->private_data; + u32 w; + + /* Reset transmitter/receiver and sample rate/frame sync generators */ + w = davinci_vcif_read_reg(dev, DAVINCI_VCIF_CTRL_REG); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + MOD_REG_BIT(w, DAVINCI_VCIF_CTRL_RSTDAC, 0); + else + MOD_REG_BIT(w, DAVINCI_VCIF_CTRL_RSTADC, 0); + + davinci_vcif_write_reg(dev, DAVINCI_VCIF_CTRL_REG, w); +} + +static int davinci_vcif_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct davinci_vcif_dev *dev = cpu_dai->private_data; + + cpu_dai->dma_data = dev->dma_params[substream->stream]; + return 0; +} + +static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct davinci_pcm_dma_params *dma_params = dai->dma_data; + struct davinci_vcif_dev *dev = dai->private_data; + u32 w; + + /* Restart the codec before setup */ + davinci_vcif_stop(substream); + davinci_vcif_start(substream); + + /* General line settings */ + davinci_vcif_write_reg(dev, + DAVINCI_VCIF_CTRL_REG, DAVINCI_VCIF_CTRL_MASK); + + davinci_vcif_write_reg(dev, + DAVINCI_VCIF_INTCLR_REG, DAVINCI_VCIF_INT_MASK); + + davinci_vcif_write_reg(dev, + DAVINCI_VCIF_INTEN_REG, DAVINCI_VCIF_INT_MASK); + + w = davinci_vcif_read_reg(dev, DAVINCI_VCIF_CTRL_REG); + + /* Determine xfer data type */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + dma_params->data_type = 0; + + MOD_REG_BIT(w, DAVINCI_VCIF_CTRL_RD_BITS_8 | + DAVINCI_VCIF_CTRL_RD_UNSIGNED | + DAVINCI_VCIF_CTRL_WD_BITS_8 | + DAVINCI_VCIF_CTRL_WD_UNSIGNED, 1); + break; + case SNDRV_PCM_FORMAT_S8: + dma_params->data_type = 1; + + MOD_REG_BIT(w, DAVINCI_VCIF_CTRL_RD_BITS_8 | + DAVINCI_VCIF_CTRL_WD_BITS_8, 1); + + MOD_REG_BIT(w, DAVINCI_VCIF_CTRL_RD_UNSIGNED | + DAVINCI_VCIF_CTRL_WD_UNSIGNED, 0); + break; + case SNDRV_PCM_FORMAT_S16_LE: + dma_params->data_type = 2; + + MOD_REG_BIT(w, DAVINCI_VCIF_CTRL_RD_BITS_8 | + DAVINCI_VCIF_CTRL_RD_UNSIGNED | + DAVINCI_VCIF_CTRL_WD_BITS_8 | + DAVINCI_VCIF_CTRL_WD_UNSIGNED, 0); + break; + default: + printk(KERN_WARNING "davinci-vcif: unsupported PCM format"); + return -EINVAL; + } + + dma_params->acnt = dma_params->data_type; + + davinci_vcif_write_reg(dev, DAVINCI_VCIF_CTRL_REG, w); + + return 0; +} + +static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + davinci_vcif_start(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_vcif_stop(substream); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +#define DAVINCI_VCIF_RATES SNDRV_PCM_RATE_8000_48000 + +static struct snd_soc_dai_ops davinci_vcif_dai_ops = { + .startup = davinci_vcif_startup, + .trigger = davinci_vcif_trigger, + .hw_params = davinci_vcif_hw_params, +}; + +struct snd_soc_dai davinci_vcif_dai = { + .name = "davinci-vcif", + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = DAVINCI_VCIF_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = DAVINCI_VCIF_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &davinci_vcif_dai_ops, + +}; +EXPORT_SYMBOL_GPL(davinci_vcif_dai); + +static int davinci_vcif_probe(struct platform_device *pdev) +{ + struct davinci_vcif_dev *dev; + struct resource *res, *mem; + int ret; + + dev = kzalloc(sizeof(struct davinci_vcif_dev), GFP_KERNEL); + if (!dev) { + dev_dbg(&pdev->dev, "could not allocate memory for private data\n"); + return -ENOMEM; + } + + dev->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(dev->clk)) { + dev_dbg(&pdev->dev, "%s: could not get the clock for voice codec\n", + pdev->name); + ret = -ENODEV; + goto fail1; + } + clk_enable(dev->clk); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "no mem resource\n"); + ret = -ENODEV; + goto fail2; + } + + dev->pbase = res->start; + dev->base_size = resource_size(res); + + mem = request_mem_region(dev->pbase, dev->base_size, pdev->name); + if (!mem) { + dev_err(&pdev->dev, "VCIF region already claimed\n"); + ret = -EBUSY; + goto fail2; + } + + dev->base = ioremap(dev->pbase, dev->base_size); + if (!dev->base) { + dev_err(&pdev->dev,"%s: can't ioremap mem resource.\n", pdev->name); + ret = -ENOMEM; + goto fail3; + } + + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &davinci_vcif_pcm_out; + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->dma_addr = + (dma_addr_t)(io_v2p(dev->base) + DAVINCI_VCIF_WFIFO_REG); + + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &davinci_vcif_pcm_in; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->dma_addr = + (dma_addr_t)(io_v2p(dev->base) + DAVINCI_VCIF_RFIFO_REG); + + /* first TX, then RX */ + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(&pdev->dev, "%s: no DMA resource\n", pdev->name); + ret = -ENXIO; + goto fail4; + } + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = res->start; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(&pdev->dev, "%s: no DMA resource\n", pdev->name); + ret = -ENXIO; + goto fail4; + } + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = res->start; + + davinci_vcif_dai.private_data = dev; + ret = snd_soc_register_dai(&davinci_vcif_dai); + if (ret != 0) + goto fail4; + + return 0; + +fail4: + iounmap(dev->base); +fail3: + release_mem_region(dev->pbase, dev->base_size); +fail2: + clk_disable(dev->clk); + clk_put(dev->clk); + dev->clk = NULL; +fail1: + kfree(dev); + + return ret; +} + +static int davinci_vcif_remove(struct platform_device *pdev) +{ + struct davinci_vcif_dev *dev = davinci_vcif_dai.private_data; + + iounmap(dev->base); + release_mem_region(dev->pbase, dev->base_size); + + clk_disable(dev->clk); + clk_put(dev->clk); + dev->clk = NULL; + + kfree(dev); + + snd_soc_unregister_dai(&davinci_vcif_dai); + + return 0; +} + +static struct platform_driver davinci_vcif_driver = { + .probe = davinci_vcif_probe, + .remove = davinci_vcif_remove, + .driver = { + .name = "voice_codec", + .owner = THIS_MODULE, + }, +}; + +static int __init davinci_vcif_init(void) +{ + return platform_driver_probe(&davinci_vcif_driver, davinci_vcif_probe); +} +module_init(davinci_vcif_init); + +static void __exit davinci_vcif_exit(void) +{ + platform_driver_unregister(&davinci_vcif_driver); +} +module_exit(davinci_vcif_exit); + +MODULE_AUTHOR("Miguel Aguilar"); +MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC Voice Codec Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/davinci/davinci-vcif.h b/sound/soc/davinci/davinci-vcif.h new file mode 100644 index 0000000..30e441d --- /dev/null +++ b/sound/soc/davinci/davinci-vcif.h @@ -0,0 +1,63 @@ +/* + * ALSA SoC Voice Codec Interface for TI DAVINCI processor + * + * Copyright (C) 2009 Texas Instruments. + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * Initial code: Hui Geng + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _DAVINCI_VCIF_H +#define _DAVINCI_VCIF_H + +#define DAVINCI_VCIF_PID_REG 0x00 +#define DAVINCI_VCIF_CTRL_REG 0x04 +#define DAVINCI_VCIF_INTEN_REG 0x08 +#define DAVINCI_VCIF_INTSTATUS_REG 0x0c +#define DAVINCI_VCIF_INTCLR_REG 0x10 +#define DAVINCI_VCIF_EMUL_CTRL_REG 0x14 +#define DAVINCI_VCIF_RFIFO_REG 0x20 +#define DAVINCI_VCIF_WFIFO_REG 0x24 +#define DAVINCI_VCIF_FIFOSTAT_REG 0x28 +#define DAVINCI_VCIF_TST_CTRL_REG 0x2C + +#define DAVINCI_VCIF_CTRL_MASK 0x5500 +#define DAVINCI_VCIF_CTRL_RSTADC (1 << 0) +#define DAVINCI_VCIF_CTRL_RSTDAC (1 << 1) +#define DAVINCI_VCIF_CTRL_RD_BITS_8 (1 << 4) +#define DAVINCI_VCIF_CTRL_RD_UNSIGNED (1 << 5) +#define DAVINCI_VCIF_CTRL_WD_BITS_8 (1 << 6) +#define DAVINCI_VCIF_CTRL_WD_UNSIGNED (1 << 7) +#define DAVINCI_VCIF_CTRL_RFIFOEN (1 << 8) +#define DAVINCI_VCIF_CTRL_RFIFOCL (1 << 9) +#define DAVINCI_VCIF_CTRL_RFIFOMD_WORD_1 (1 << 10) +#define DAVINCI_VCIF_CTRL_WFIFOEN (1 << 12) +#define DAVINCI_VCIF_CTRL_WFIFOCL (1 << 13) +#define DAVINCI_VCIF_CTRL_WFIFOMD_WORD_1 (1 << 14) + +#define DAVINCI_VCIF_INT_MASK 0x3F +#define DAVINCI_VCIF_INT_RDRDY_MASK (1 << 0) +#define DAVINCI_VCIF_INT_RERROVF_MASK (1 << 1) +#define DAVINCI_VCIF_INT_RERRUDR_MASK (1 << 2) +#define DAVINCI_VCIF_INT_WDREQ_MASK (1 << 3) +#define DAVINCI_VCIF_INT_WERROVF_MASK (1 << 4) +#define DAVINCI_VCIF_INT_WERRUDR_MASK (1 << 5) + +extern struct snd_soc_dai davinci_vcif_dai; + +#endif -- 1.6.0.4

3 7

Re: [alsa-devel] Testers wanted: New SB X-Fi driver
by Christian Esken 18 Nov '09

18 Nov '09

Am Sonntag 12 Juli 2009 10:24:35 schrieb Christian Esken: > Am Montag 06 Juli 2009 08:26:11 schrieb Takashi Iwai: > > At Sun, 5 Jul 2009 13:29:48 +0200, > > > > Christian Esken wrote: > > > Am Sonntag 05 Juli 2009 09:44:33 schrieb Takashi Iwai: > > > > At Sun, 5 Jul 2009 00:19:55 +0200, > > > > > > > > Christian Esken wrote: > > > > > Am Samstag 27 Juni 2009 14:05:45 schrieb Christian Esken: > > > > > > Hello, > > > > > > > > > > > > unfortunately I cannot playback anything (PCM) with the new SB > > > > > > X-Fi driver. Driver: > > > > > > ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa- > > > > > > driver-snapshot.tar.bz2 > > > > > > Distribution: openSuSE 11.1 > > > > > > Kernel: 2.6.27.23-0.1-xen > > > > > > > > > > > > My card is a "X-Fi Titanium" (PCI device code: 1102:000b). The > > > > > > card is recognized by the driver, and any mixer (alsamixer, kmix) > > > > > > also recognizes the card. > > > > > > > > > > > > But playing back isn't possible. I tried speaker-test, audacity, > > > > > > amarok and aplay. Most applications hang, some also produce an > > > > > > error in the /var/log/messages (see > > > > > > below). > > > > > > > > > > > > I then used "./configure --with-debug=full --with-cards=ctxfi" > > > > > > due to the possible hd-audio conflict, but it didn't make a > > > > > > difference. > > > > > > > > > > > > Is there anything else I could try? > > > > > > > > > > I tested the same openSuSE version with a non-Xen 64 bit kernel, > > > > > and it doesn't work either. > > > > > > > > > > And I tried with this: > > > > > Driver: alsa-driver-20090704.tar.bz2 > > > > > Distribution: Kubuntu Karmic Alpha-2 > > > > > Kernel: 2.6.30 (32 Bit) > > > > > > > > > > Under the latter setup the X-Fi Titanium works (Simple 2 channel > > > > > test with speaker-test, audacity, amarok). :-)) > > > > > > > > How many RAM do you have? > > > > Could you boot with the limited memory size? > > > > > > It is just 2GB. So I am not limited by the 32 bit barrier. > > > > > > So booting with "limited" memory size doesn't make sense, right? > > > > Supposedly. Is it emu20k1 or emu20k2? > > I checked it again: It is a 20k2 chip. > > /var/log/boot.msg says: > <7>ALSA /home/chris/Desktop/alsa-driver/pci/ctxfi/ctatc.c:1268: ctxfi: chip > 20K2 model SB0880 (1102:0041) is found > > > http://pci-ids.ucw.cz/mods/PC/1102/000b shows 1102:0041 as "X-Fi Titanium > series [EMU20k2]". I did now test it with a different 64 Bit Linux distribution, and it works correctly: Driver: alsa-driver-20090712.tar.bz2 Distribution: Kubuntu Karmic Alpha-2 Kernel: 2.6.30 (64 Bit) All in all, I can produce a working environment with recent 32 and 64 bit kernels. This is good news. :-) Below is the summary on the tested distributions. Christian So we got this: Kubuntu Karmic Alpha-2 32 Bit, 2.0.30 (works) Kubuntu Karmic Alpha-2 64 Bit, 2.0.30 (works) SuSE 11.1 64 Bit, 2.0.27 (does not work) SuSE 11.1 64 Bit, 2.0.27 XEN (does not work) SuSE 11.2 Milestone3 64Bit, 2.0.30 (test canceled, due to unrelated problems) => I'll see whether I can redo the SuSE11.2 tests, after doing a bugreport at openSuSE.

3 6

[alsa-devel] [RFC][PATCH] ARM: OMAP: McBSP: Use register cache
by Janusz Krzysztofik 14 Nov '09

14 Nov '09

Change the way McBSP registers are updated: use cached values instead of relying upon those read back from the device. With this patch, I have finally managed to get rid of all random playback/recording hangups on my OMAP1510 based Amstrad Delta (buggy?) hardware. Before that, I could suspect that values read back from McBSP registers before updating them happened to be errornous. I think there is one important point that makes this patch worth of applying, apart from my hardware quality. With the current code, if it ever happens to any machine, no matter if OMAP1510 or newer, to read incorrect value from a McBSP register, this wrong value will get written back without any checking. That can lead to hardware damage if, for example, an input pin is turned into output as a result. Created against linux-2.6.31-rc5 Tested on Amstrad Delta Signed-off-by: Janusz Krzysztofik --- --- linux-2.6.31-rc5/arch/arm/plat-omap/include/mach/mcbsp.h.orig 2009-08-11 23:43:25.000000000 +0200 +++ linux-2.6.31-rc5/arch/arm/plat-omap/include/mach/mcbsp.h 2009-08-11 23:45:46.000000000 +0200 @@ -377,6 +377,8 @@ struct omap_mcbsp { struct omap_mcbsp_platform_data *pdata; struct clk *iclk; struct clk *fclk; + + struct omap_mcbsp_reg_cfg reg_cache; }; extern struct omap_mcbsp **mcbsp_ptr; extern int omap_mcbsp_count; --- linux-2.6.31-rc5/arch/arm/plat-omap/mcbsp.c.orig 2009-08-11 23:43:25.000000000 +0200 +++ linux-2.6.31-rc5/arch/arm/plat-omap/mcbsp.c 2009-08-11 23:45:35.000000000 +0200 @@ -91,6 +91,7 @@ static void omap_mcbsp_dump_reg(u8 id) static irqreturn_t omap_mcbsp_tx_irq_handler(int irq, void *dev_id) { struct omap_mcbsp *mcbsp_tx = dev_id; + struct omap_mcbsp_reg_cfg *reg_cache = &mcbsp_tx->reg_cache; u16 irqst_spcr2; irqst_spcr2 = OMAP_MCBSP_READ(mcbsp_tx->io_base, SPCR2); @@ -101,7 +102,7 @@ static irqreturn_t omap_mcbsp_tx_irq_han irqst_spcr2); /* Writing zero to XSYNC_ERR clears the IRQ */ OMAP_MCBSP_WRITE(mcbsp_tx->io_base, SPCR2, - irqst_spcr2 & ~(XSYNC_ERR)); + reg_cache->spcr2 &= ~(XSYNC_ERR)); } else { complete(&mcbsp_tx->tx_irq_completion); } @@ -112,6 +113,7 @@ static irqreturn_t omap_mcbsp_tx_irq_han static irqreturn_t omap_mcbsp_rx_irq_handler(int irq, void *dev_id) { struct omap_mcbsp *mcbsp_rx = dev_id; + struct omap_mcbsp_reg_cfg *reg_cache = &mcbsp_rx->reg_cache; u16 irqst_spcr1; irqst_spcr1 = OMAP_MCBSP_READ(mcbsp_rx->io_base, SPCR1); @@ -122,7 +124,7 @@ static irqreturn_t omap_mcbsp_rx_irq_han irqst_spcr1); /* Writing zero to RSYNC_ERR clears the IRQ */ OMAP_MCBSP_WRITE(mcbsp_rx->io_base, SPCR1, - irqst_spcr1 & ~(RSYNC_ERR)); + reg_cache->spcr1 &= ~(RSYNC_ERR)); } else { complete(&mcbsp_rx->tx_irq_completion); } @@ -167,6 +169,7 @@ static void omap_mcbsp_rx_dma_callback(i void omap_mcbsp_config(unsigned int id, const struct omap_mcbsp_reg_cfg *config) { struct omap_mcbsp *mcbsp; + struct omap_mcbsp_reg_cfg *reg_cache; void __iomem *io_base; if (!omap_mcbsp_check_valid_id(id)) { @@ -174,26 +177,27 @@ void omap_mcbsp_config(unsigned int id, return; } mcbsp = id_to_mcbsp_ptr(id); + reg_cache = &mcbsp->reg_cache; io_base = mcbsp->io_base; dev_dbg(mcbsp->dev, "Configuring McBSP%d phys_base: 0x%08lx\n", mcbsp->id, mcbsp->phys_base); /* We write the given config */ - OMAP_MCBSP_WRITE(io_base, SPCR2, config->spcr2); - OMAP_MCBSP_WRITE(io_base, SPCR1, config->spcr1); - OMAP_MCBSP_WRITE(io_base, RCR2, config->rcr2); - OMAP_MCBSP_WRITE(io_base, RCR1, config->rcr1); - OMAP_MCBSP_WRITE(io_base, XCR2, config->xcr2); - OMAP_MCBSP_WRITE(io_base, XCR1, config->xcr1); - OMAP_MCBSP_WRITE(io_base, SRGR2, config->srgr2); - OMAP_MCBSP_WRITE(io_base, SRGR1, config->srgr1); - OMAP_MCBSP_WRITE(io_base, MCR2, config->mcr2); - OMAP_MCBSP_WRITE(io_base, MCR1, config->mcr1); - OMAP_MCBSP_WRITE(io_base, PCR0, config->pcr0); + OMAP_MCBSP_WRITE(io_base, SPCR2, reg_cache->spcr2 = config->spcr2); + OMAP_MCBSP_WRITE(io_base, SPCR1, reg_cache->spcr1 = config->spcr1); + OMAP_MCBSP_WRITE(io_base, RCR2, reg_cache->rcr2 = config->rcr2); + OMAP_MCBSP_WRITE(io_base, RCR1, reg_cache->rcr1 = config->rcr1); + OMAP_MCBSP_WRITE(io_base, XCR2, reg_cache->xcr2 = config->xcr2); + OMAP_MCBSP_WRITE(io_base, XCR1, reg_cache->xcr1 = config->xcr1); + OMAP_MCBSP_WRITE(io_base, SRGR2, reg_cache->srgr2 = config->srgr2); + OMAP_MCBSP_WRITE(io_base, SRGR1, reg_cache->srgr1 = config->srgr1); + OMAP_MCBSP_WRITE(io_base, MCR2, reg_cache->mcr2 = config->mcr2); + OMAP_MCBSP_WRITE(io_base, MCR1, reg_cache->mcr1 = config->mcr1); + OMAP_MCBSP_WRITE(io_base, PCR0, reg_cache->pcr0 = config->pcr0); if (cpu_is_omap2430() || cpu_is_omap34xx()) { - OMAP_MCBSP_WRITE(io_base, XCCR, config->xccr); - OMAP_MCBSP_WRITE(io_base, RCCR, config->rccr); + OMAP_MCBSP_WRITE(io_base, XCCR, reg_cache->xccr = config->xccr); + OMAP_MCBSP_WRITE(io_base, RCCR, reg_cache->rccr = config->rccr); } } EXPORT_SYMBOL(omap_mcbsp_config); @@ -232,6 +236,7 @@ EXPORT_SYMBOL(omap_mcbsp_set_io_type); int omap_mcbsp_request(unsigned int id) { struct omap_mcbsp *mcbsp; + struct omap_mcbsp_reg_cfg *reg_cache; int err; if (!omap_mcbsp_check_valid_id(id)) { @@ -239,6 +244,7 @@ int omap_mcbsp_request(unsigned int id) return -ENODEV; } mcbsp = id_to_mcbsp_ptr(id); + reg_cache = &mcbsp->reg_cache; spin_lock(&mcbsp->lock); if (!mcbsp->free) { @@ -261,8 +267,8 @@ int omap_mcbsp_request(unsigned int id) * Make sure that transmitter, receiver and sample-rate generator are * not running before activating IRQs. */ - OMAP_MCBSP_WRITE(mcbsp->io_base, SPCR1, 0); - OMAP_MCBSP_WRITE(mcbsp->io_base, SPCR2, 0); + OMAP_MCBSP_WRITE(mcbsp->io_base, SPCR1, reg_cache->spcr1 = 0); + OMAP_MCBSP_WRITE(mcbsp->io_base, SPCR2, reg_cache->spcr2 = 0); if (mcbsp->io_type == OMAP_MCBSP_IRQ_IO) { /* We need to get IRQs here */ @@ -335,42 +341,38 @@ EXPORT_SYMBOL(omap_mcbsp_free); void omap_mcbsp_start(unsigned int id, int tx, int rx) { struct omap_mcbsp *mcbsp; + struct omap_mcbsp_reg_cfg *reg_cache; void __iomem *io_base; int idle; - u16 w; if (!omap_mcbsp_check_valid_id(id)) { printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); return; } mcbsp = id_to_mcbsp_ptr(id); + reg_cache = &mcbsp->reg_cache; io_base = mcbsp->io_base; - mcbsp->rx_word_length = (OMAP_MCBSP_READ(io_base, RCR1) >> 5) & 0x7; - mcbsp->tx_word_length = (OMAP_MCBSP_READ(io_base, XCR1) >> 5) & 0x7; + mcbsp->rx_word_length = (reg_cache->rcr1 >> 5) & 0x7; + mcbsp->tx_word_length = (reg_cache->xcr1 >> 5) & 0x7; - idle = !((OMAP_MCBSP_READ(io_base, SPCR2) | - OMAP_MCBSP_READ(io_base, SPCR1)) & 1); + idle = !((reg_cache->spcr2 | reg_cache->spcr1) & 1); if (idle) { /* Start the sample generator */ - w = OMAP_MCBSP_READ(io_base, SPCR2); - OMAP_MCBSP_WRITE(io_base, SPCR2, w | (1 << 6)); + OMAP_MCBSP_WRITE(io_base, SPCR2, reg_cache->spcr2 |= (1 << 6)); } /* Enable transmitter and receiver */ - w = OMAP_MCBSP_READ(io_base, SPCR2); - OMAP_MCBSP_WRITE(io_base, SPCR2, w | (tx & 1)); + OMAP_MCBSP_WRITE(io_base, SPCR2, reg_cache->spcr2 |= (tx & 1)); - w = OMAP_MCBSP_READ(io_base, SPCR1); - OMAP_MCBSP_WRITE(io_base, SPCR1, w | (rx & 1)); + OMAP_MCBSP_WRITE(io_base, SPCR1, reg_cache->spcr1 |= (rx & 1)); udelay(100); if (idle) { /* Start frame sync */ - w = OMAP_MCBSP_READ(io_base, SPCR2); - OMAP_MCBSP_WRITE(io_base, SPCR2, w | (1 << 7)); + OMAP_MCBSP_WRITE(io_base, SPCR2, reg_cache->spcr2 |= (1 << 7)); } /* Dump McBSP Regs */ @@ -381,9 +383,9 @@ EXPORT_SYMBOL(omap_mcbsp_start); void omap_mcbsp_stop(unsigned int id, int tx, int rx) { struct omap_mcbsp *mcbsp; + struct omap_mcbsp_reg_cfg *reg_cache; void __iomem *io_base; int idle; - u16 w; if (!omap_mcbsp_check_valid_id(id)) { printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); @@ -391,23 +393,20 @@ void omap_mcbsp_stop(unsigned int id, in } mcbsp = id_to_mcbsp_ptr(id); + reg_cache = &mcbsp->reg_cache; io_base = mcbsp->io_base; /* Reset transmitter */ - w = OMAP_MCBSP_READ(io_base, SPCR2); - OMAP_MCBSP_WRITE(io_base, SPCR2, w & ~(tx & 1)); + OMAP_MCBSP_WRITE(io_base, SPCR2, reg_cache->spcr2 &= ~(tx & 1)); /* Reset receiver */ - w = OMAP_MCBSP_READ(io_base, SPCR1); - OMAP_MCBSP_WRITE(io_base, SPCR1, w & ~(rx & 1)); + OMAP_MCBSP_WRITE(io_base, SPCR1, reg_cache->spcr1 &= ~(rx & 1)); - idle = !((OMAP_MCBSP_READ(io_base, SPCR2) | - OMAP_MCBSP_READ(io_base, SPCR1)) & 1); + idle = !((reg_cache->spcr2 | reg_cache->spcr1) & 1); if (idle) { /* Reset the sample rate generator */ - w = OMAP_MCBSP_READ(io_base, SPCR2); - OMAP_MCBSP_WRITE(io_base, SPCR2, w & ~(1 << 6)); + OMAP_MCBSP_WRITE(io_base, SPCR2, reg_cache->spcr2 &= ~(1 << 6)); } } EXPORT_SYMBOL(omap_mcbsp_stop); @@ -557,6 +556,7 @@ EXPORT_SYMBOL(omap_mcbsp_recv_word); int omap_mcbsp_spi_master_xmit_word_poll(unsigned int id, u32 word) { struct omap_mcbsp *mcbsp; + struct omap_mcbsp_reg_cfg *reg_cache; void __iomem *io_base; omap_mcbsp_word_length tx_word_length; omap_mcbsp_word_length rx_word_length; @@ -567,6 +567,7 @@ int omap_mcbsp_spi_master_xmit_word_poll return -ENODEV; } mcbsp = id_to_mcbsp_ptr(id); + reg_cache = &mcbsp->reg_cache; io_base = mcbsp->io_base; tx_word_length = mcbsp->tx_word_length; rx_word_length = mcbsp->rx_word_length; @@ -580,9 +581,11 @@ int omap_mcbsp_spi_master_xmit_word_poll spcr2 = OMAP_MCBSP_READ(io_base, SPCR2); if (attempts++ > 1000) { /* We must reset the transmitter */ - OMAP_MCBSP_WRITE(io_base, SPCR2, spcr2 & (~XRST)); + OMAP_MCBSP_WRITE(io_base, SPCR2, + reg_cache->spcr2 &= (~XRST)); udelay(10); - OMAP_MCBSP_WRITE(io_base, SPCR2, spcr2 | XRST); + OMAP_MCBSP_WRITE(io_base, SPCR2, + reg_cache->spcr2 |= XRST); udelay(10); dev_err(mcbsp->dev, "McBSP%d transmitter not " "ready\n", mcbsp->id); @@ -601,9 +604,11 @@ int omap_mcbsp_spi_master_xmit_word_poll spcr1 = OMAP_MCBSP_READ(io_base, SPCR1); if (attempts++ > 1000) { /* We must reset the receiver */ - OMAP_MCBSP_WRITE(io_base, SPCR1, spcr1 & (~RRST)); + OMAP_MCBSP_WRITE(io_base, SPCR1, + reg_cache->spcr1 &= (~RRST)); udelay(10); - OMAP_MCBSP_WRITE(io_base, SPCR1, spcr1 | RRST); + OMAP_MCBSP_WRITE(io_base, SPCR1, + reg_cache->spcr1 |= RRST); udelay(10); dev_err(mcbsp->dev, "McBSP%d receiver not " "ready\n", mcbsp->id); @@ -623,6 +628,7 @@ EXPORT_SYMBOL(omap_mcbsp_spi_master_xmit int omap_mcbsp_spi_master_recv_word_poll(unsigned int id, u32 *word) { struct omap_mcbsp *mcbsp; + struct omap_mcbsp_reg_cfg *reg_cache; u32 clock_word = 0; void __iomem *io_base; omap_mcbsp_word_length tx_word_length; @@ -635,6 +641,7 @@ int omap_mcbsp_spi_master_recv_word_poll } mcbsp = id_to_mcbsp_ptr(id); + reg_cache = &mcbsp->reg_cache; io_base = mcbsp->io_base; tx_word_length = mcbsp->tx_word_length; @@ -649,9 +656,11 @@ int omap_mcbsp_spi_master_recv_word_poll spcr2 = OMAP_MCBSP_READ(io_base, SPCR2); if (attempts++ > 1000) { /* We must reset the transmitter */ - OMAP_MCBSP_WRITE(io_base, SPCR2, spcr2 & (~XRST)); + OMAP_MCBSP_WRITE(io_base, SPCR2, + reg_cache->spcr2 &= (~XRST)); udelay(10); - OMAP_MCBSP_WRITE(io_base, SPCR2, spcr2 | XRST); + OMAP_MCBSP_WRITE(io_base, SPCR2, + reg_cache->spcr2 |= XRST); udelay(10); dev_err(mcbsp->dev, "McBSP%d transmitter not " "ready\n", mcbsp->id); @@ -670,9 +679,11 @@ int omap_mcbsp_spi_master_recv_word_poll spcr1 = OMAP_MCBSP_READ(io_base, SPCR1); if (attempts++ > 1000) { /* We must reset the receiver */ - OMAP_MCBSP_WRITE(io_base, SPCR1, spcr1 & (~RRST)); + OMAP_MCBSP_WRITE(io_base, SPCR1, + reg_cache->spcr1 &= (~RRST)); udelay(10); - OMAP_MCBSP_WRITE(io_base, SPCR1, spcr1 | RRST); + OMAP_MCBSP_WRITE(io_base, SPCR1, + reg_cache->spcr1 |= RRST); udelay(10); dev_err(mcbsp->dev, "McBSP%d receiver not " "ready\n", mcbsp->id);

1 2

[alsa-devel] Trying to get unsupported CODEC to work
by Alex Austin 05 Nov '09

05 Nov '09

HelloI have an MSI Media Live DIVA motherboard. It has an AMD SB600 HD Audio controller, and two CODECs on the HDA bus. First is an ALC888 Second is an Intersil DAE-3. Two alsa-info outputs in different circumstances http://www.alsa-project.org/db/?f=20c86e755fdbbdbc8761c8c2f4de76bce2ea0c11 http://www.alsa-project.org/db/?f=15c5e5eab0104622a84b2dd0df03f26c17c7242d Output of codecgraph with "modprobe hda-intel probe_mask=2" http://pastebin.com/f54b6a9b8 What's the next step to get this supported? -- - Alex Austin (651) 238-9273 "...and then I visited Wikipedia ...and the next 8 hours are a blur."

3 13

[alsa-devel] [Fwd: Scratch Live amplifier switch]
by LCID Fire 02 Nov '09

02 Nov '09

Hi. Following the patch to add the amplifier switch (changing input levels phono/line) for the Serato Scratch Live device. Please apply or complain ;)

2 3

[alsa-devel] [PATCH] ASoC: OMAP3EVM: Use the twl4030_setup_data for headset pop-removal
by Anuj Aggarwal 29 Oct '09

29 Oct '09

The pop-removal specific values are configured for TWL4030 codec for OMAP3EVM through this patch. Signed-off-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> --- sound/soc/omap/omap3evm.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c index 9114c26..8deb59b 100644 --- a/sound/soc/omap/omap3evm.c +++ b/sound/soc/omap/omap3evm.c @@ -93,10 +93,17 @@ static struct snd_soc_card snd_soc_omap3evm = { .num_links = 1, }; +/* twl4030 setup */ +static struct twl4030_setup_data twl4030_setup = { + .ramp_delay_value = 4, + .sysclk = 26000, +}; + /* Audio subsystem */ static struct snd_soc_device omap3evm_snd_devdata = { .card = &snd_soc_omap3evm, .codec_dev = &soc_codec_dev_twl4030, + .codec_data = &twl4030_setup, }; static struct platform_device *omap3evm_snd_device; -- 1.6.2.4

4 3

[alsa-devel] Questions on locking, snd-usb-caiaq
by Mark Hills 18 Oct '09

18 Oct '09

I fixed some race issues in snd-usb-caiaq, but have some questions on the interface between the driver and ALSA. First fix is a lock in pointer(), which caused snd_pcm_update_hw_ptr_pos() to report an out of range buffer position (even though it was sucessfully corrected to zero). Second is a lock as part of trigger(). On testing, this appears to fix an issue where white noise could sometimes be transferred by the driver. According to the documentation, trigger() and pointer() are called with local interrupts disabled. This means they should really be non-irqsave locks? (http://www.alsa-project.org/~tiwai/writing-an-alsa-driver/ch05s08.html) Even for callbacks documented as atomic, if read_completed() and write_completed() can be called at any time interrupts are enabled (eg. on another CPU) it seems that additional locking my be required. Candidates are: snd_usb_caiaq_substream_open snd_usb_caiaq_substream_close snd_usb_caiaq_pcm_hw_free snd_usb_caiaq_pcm_prepare Are there any guarantees on these functions which mean the lock is not required? To lock in prepare() would (I think) require demoting a spin_lock_irqsave() to a spin_lock() to send initialisation commands. If so, what is the best design for this? Thanks, -- Mark diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 121af06..c659f81 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -62,10 +62,15 @@ static void activate_substream(struct snd_usb_caiaqdev *dev, struct snd_pcm_substream *sub) { + unsigned long flags; + spin_lock_irqsave(&dev->spinlock, flags); + if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) dev->sub_playback[sub->number] = sub; else dev->sub_capture[sub->number] = sub; + + spin_unlock_irqrestore(&dev->spinlock, flags); } static void @@ -269,16 +274,23 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub) { int index = sub->number; struct snd_usb_caiaqdev *dev = snd_pcm_substream_chip(sub); + unsigned long flags; + snd_pcm_uframes_t ptr; + + spin_lock_irqsave(&dev->spinlock, flags); if (dev->input_panic || dev->output_panic) - return SNDRV_PCM_POS_XRUN; + ptr = SNDRV_PCM_POS_XRUN; if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) - return bytes_to_frames(sub->runtime, + ptr = bytes_to_frames(sub->runtime, dev->audio_out_buf_pos[index]); else - return bytes_to_frames(sub->runtime, + ptr = bytes_to_frames(sub->runtime, dev->audio_in_buf_pos[index]); + + spin_unlock_irqrestore(&dev->spinlock, flags); + return ptr; } /* operators for both playback and capture */

3 12

[alsa-devel] [PATCHv2 5/7] ASoC: TWL6030: Add support for low-power mode
by Lopez Cruz, Misael 14 Oct '09

14 Oct '09

TWL6030 codec supports two power modes: low-power and high performance. In low-power mode, headset downlink must be the only path enabled and components in that path (headset DAC and driver) should be in that mode too. In this mode, codec can stream audio at 44.1 and 48 kHz if sys clock from CLK32K is configured to 17.64 and 19.2 MHz using low-power PLL, respectively. In high-performance mode, codec can only work at 19.2 MHz from high-performance PLL. All paths supported in the codec can be used but audio can be streamed only at 48 kHz. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl6030.c | 214 ++++++++++++++++++++++++++++++++++++++------ sound/soc/codecs/twl6030.h | 17 ++++ 2 files changed, 202 insertions(+), 29 deletions(-) diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c index f1e333f..032619d 100644 --- a/sound/soc/codecs/twl6030.c +++ b/sound/soc/codecs/twl6030.c @@ -46,6 +46,7 @@ struct twl6030_data { struct snd_soc_codec codec; int codec_powered; + unsigned int sysclk; }; /* @@ -305,6 +306,29 @@ static void twl6030_power_down(struct snd_soc_codec *codec) mdelay(10); } +/* set headset dac and driver power mode */ +static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) +{ + int hslctl, hsrctl; + int mask = TWL6030_HSDRVMODEL | TWL6030_HSDACMODEL; + + hslctl = twl6030_read_reg_cache(codec, TWL6030_REG_HSLCTL); + hsrctl = twl6030_read_reg_cache(codec, TWL6030_REG_HSRCTL); + + if (high_perf) { + hslctl &= ~mask; + hsrctl &= ~mask; + } else { + hslctl |= mask; + hsrctl |= mask; + } + + twl6030_write(codec, TWL6030_REG_HSLCTL, hslctl); + twl6030_write(codec, TWL6030_REG_HSRCTL, hsrctl); + + return 0; +} + /* * MICATT volume control: * from -6 to 0 dB in 6 dB steps @@ -598,6 +622,7 @@ static int twl6030_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; + struct twl6030_data *priv = codec->private_data; int rate, format; /* hardware dai (McPDM) requires bit stream of twice @@ -607,7 +632,20 @@ static int twl6030_hw_params(struct snd_pcm_substream *substream, rate = params_rate(params); switch (rate) { case 44100: + if (priv->sysclk != 17640000) { + dev_err(codec->dev, + "rate %d not supported at current sysclk %d\n", + rate, priv->sysclk); + return -EINVAL; + } + break; case 48000: + if (priv->sysclk != 19200000) { + dev_err(codec->dev, + "rate %d not supported at current sysclk %d\n", + rate, priv->sysclk); + return -EINVAL; + } break; default: dev_err(codec->dev, "unknown rate %d\n", rate); @@ -630,46 +668,163 @@ static int twl6030_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; + struct twl6030_data *priv = codec->private_data; u8 hppll, lppll; + lppll = twl6030_read_reg_cache(codec, TWL6030_REG_LPPLLCTL); hppll = twl6030_read_reg_cache(codec, TWL6030_REG_HPPLLCTL); - hppll &= TWL6030_HPLLRST; - - switch (freq) { - case 12000000: - /* MCLK input, PLL enabled */ - hppll = TWL6030_MCLK_12000KHZ - | TWL6030_HPLLSQRBP - | TWL6030_HPLLENA; - break; - case 19200000: - /* MCLK input, PLL disabled */ - hppll = TWL6030_MCLK_19200KHZ - | TWL6030_HPLLSQRBP - | TWL6030_HPLLBP; + + switch (clk_id) { + case TWL6030_SYSCLK_SEL_LPPLL: + if (freq != 32768) { + dev_err(codec->dev, "invalid sysclk freq %d\n", freq); + return -EINVAL; + } + + /* CLK32K input requires low-power PLL */ + lppll |= TWL6030_LPLLENA | TWL6030_LPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + mdelay(5); + lppll &= ~TWL6030_HPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + + /* headset dac and driver must be in low-power mode */ + headset_power_mode(codec, 0); + + hppll &= ~TWL6030_HPLLENA; + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppll); break; - case 26000000: - /* MCLK input, PLL enabled */ - hppll = TWL6030_MCLK_26000KHZ - | TWL6030_HPLLSQRBP - | TWL6030_HPLLENA; + case TWL6030_SYSCLK_SEL_HPPLL: + switch (freq) { + case 12000000: + hppll = TWL6030_MCLK_12000KHZ; + break; + case 26000000: + hppll = TWL6030_MCLK_26000KHZ; + break; + default: + dev_err(codec->dev, "invalid sysclk freq %d\n", freq); + return -EINVAL; + } + + /* 12 and 26 MHz freqs require high-performance PLL */ + hppll |= TWL6030_HPLLSQRBP | TWL6030_HPLLENA; + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppll); + udelay(500); + + /* headset dac and driver must be in high-performance mode */ + headset_power_mode(codec, 1); + + lppll |= TWL6030_HPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + lppll &= ~TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + + /* only 19.2 MHz can be generated by HPPLL */ + priv->sysclk = 19200000; break; - case 38400000: - /* clk slicer input, PLL disabled */ - hppll = TWL6030_MCLK_38400KHZ - | TWL6030_HPLLSQRENA - | TWL6030_HPLLBP; + case TWL6030_SYSCLK_SEL_MCLK: + switch (freq) { + case 19200000: + hppll = TWL6030_MCLK_19200KHZ | TWL6030_HPLLSQRBP; + break; + case 38400000: + hppll = TWL6030_MCLK_38400KHZ; + break; + default: + dev_err(codec->dev, "invalid sysclk freq %d\n", freq); + return -EINVAL; + } + + /* 19.2 and 38.4 MHz freqs don't require PLL */ + hppll |= TWL6030_HPLLBP; + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppll); + udelay(500); + + /* headset dac and driver must be in high-performance mode */ + headset_power_mode(codec, 1); + + lppll |= TWL6030_HPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + lppll &= ~TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + + /* only 19.2 MHz can be generated by MCLK */ + priv->sysclk = 19200000; break; default: - dev_err(codec->dev, "unknown sysclk rate %d\n", freq); + dev_err(codec->dev, "unknown clk_id %d\n", clk_id); return -EINVAL; } - twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppll); + return 0; +} + +static int twl6030_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct twl6030_data *priv = codec->private_data; + int div; + u8 ldoctl, lppllctl, hppllctl; + + ldoctl = twl6030_read_reg_cache(codec, TWL6030_REG_LDOCTL); + lppllctl = twl6030_read_reg_cache(codec, TWL6030_REG_LPPLLCTL); + hppllctl = twl6030_read_reg_cache(codec, TWL6030_REG_HPPLLCTL); + + if (freq_in && freq_out) { + /* div = round(freq_out / freq_in) */ + div = (freq_out + (freq_in >> 2)) / freq_in; + + switch (pll_id) { + case TWL6030_LPPLL_ID: + if (div < 512) { + dev_err(codec->dev, "invalid pll div %d\n", + div); + return -EINVAL; + } + twl6030_write(codec, TWL6030_REG_LPPLLDIV, div - 512); + lppllctl |= TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + + priv->sysclk = freq_out; + break; + case TWL6030_HPPLL_ID: + hppllctl |= TWL6030_HPLLENA; + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppllctl); + break; + default: + dev_err(codec->dev, "unknown pll id %d\n", pll_id); + return -EINVAL; + } - /* Disable LPPLL and select HPPLL */ - lppll = TWL6030_HPLLSEL; - twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + /* switch to pll */ + ldoctl &= ~TWL6030_OSCENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(10); + } else { + /* switch to internal oscillator */ + ldoctl |= TWL6030_OSCENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(10); + + /* turn-off pll */ + switch (pll_id) { + case TWL6030_HPPLL_ID: + hppllctl &= ~TWL6030_HPLLENA; + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppllctl); + break; + case TWL6030_LPPLL_ID: + lppllctl &= ~TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + break; + default: + dev_err(codec->dev, "unknown pll id %d\n", pll_id); + return -EINVAL; + } + + priv->sysclk = 0; + } return 0; } @@ -689,6 +844,7 @@ static int twl6030_set_dai_fmt(struct snd_soc_dai *codec_dai, static struct snd_soc_dai_ops twl6030_dai_ops = { .hw_params = twl6030_hw_params, .set_sysclk = twl6030_set_dai_sysclk, + .set_pll = twl6030_set_dai_pll, .set_fmt = twl6030_set_dai_fmt, }; diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h index 7375ae8..15d3e1b 100644 --- a/sound/soc/codecs/twl6030.h +++ b/sound/soc/codecs/twl6030.h @@ -92,6 +92,10 @@ #define TWL6030_MCLK_38400KHZ (3 << 5) #define TWL6030_MCLK_MSK 0x60 +#define TWL6030_SYSCLK_SEL_LPPLL 1 +#define TWL6030_SYSCLK_SEL_HPPLL 2 +#define TWL6030_SYSCLK_SEL_MCLK 3 + /* LPPLLCTL (0x08) fields */ #define TWL6030_LPLLENA 0x01 @@ -100,6 +104,19 @@ #define TWL6030_LPLLFIN 0x08 #define TWL6030_HPLLSEL 0x10 +#define TWL6030_HPPLL_ID 1 +#define TWL6030_LPPLL_ID 2 + +/* HSLCTL (0x10) fields */ + +#define TWL6030_HSDACMODEL 0x02 +#define TWL6030_HSDRVMODEL 0x08 + +/* HSRCTL (0x11) fields */ + +#define TWL6030_HSDACMODER 0x02 +#define TWL6030_HSDRVMODER 0x08 + /* ACCCTL (0x2D) fields */ #define TWL6030_RESETSPLIT 0x04 -- 1.5.4.3

3 6

[alsa-devel] [PATCH] snd-usb-us122l: added support for US-144
by Tobias Hansen 13 Oct '09

13 Oct '09

Hi, here is a patch that adds support for the Tascam US-144 to snd-usb-us122l when uhci-hcd is used (ehci-hcd has to be disabled). With ehci-hcd the US-144 behaves completely differently. The difference between US-122L and US-144 on uhci-hcd is, that US-144 uses endpoints on two interfaces, while on US-122L all endpoints are on one interface. US-122L should still work, but I don't have a US-122L to test that. Audio in and out and Midi in were tested with US-144 and work. This is my first contribution to the linux kernel, so please have a critical look on it. ;) Taken against snd-usb-us122l 0.5 / alsa 1.0.20. Regards, Tobias Hansen

2 7

Re: [alsa-devel] [PATCH REALTEK] Added quirk to enable sound on Toshiba NB200
by Takashi Iwai 08 Oct '09

08 Oct '09

At Wed, 23 Sep 2009 00:39:45 -0500 (CDT), Manoj Iyer wrote: > > > >From c96629e96fecbe59de7124ec913019ed378d0029 Mon Sep 17 00:00:00 2001 > From: Manoj Iyer <manoj.iyer(a)canonical.com> > Date: Tue, 22 Sep 2009 18:33:29 -0500 > Subject: [PATCH] [PATCH REALTEK] Added quirk to enable sound on Toshiba NB200 > > Patch was tested on Toshiba NB200 and is found to enable sound. So, it doesn't work without the model quirk? Could you give alsa-info.sh output (run with --no-upload option)? thanks, Takashi > > Signed-off-by: Manoj Iyer <manoj.iyer(a)canonical.com> > Cc: stable(a)kernel.org > --- > sound/pci/hda/patch_realtek.c | 1 + > 1 files changed, 1 insertions(+), 0 deletions(-) > > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index 30eeb30..713bf69 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -16876,6 +16876,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { > SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS), > SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K", > ALC662_3ST_6ch_DIG), > + SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB200", ALC663_ASUS_MODE4), > SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10), > SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L", > ALC662_3ST_6ch_DIG), > -- > 1.6.3.3 > > > --- manjo >

2 13

[alsa-devel] Application supporting forward/rewind APIs
by Anuj Aggarwal 07 Oct '09

07 Oct '09

Hi, I want to try the forward/rewind APIs supported by the ALSA library. Can someone point to an ALSA application which exercises these APIs? Best Regards, Anuj Aggarwal

4 8

[alsa-devel] hda_intel: SPDIF not detected on Asus P7P55D PRO
by Robert Hancock 06 Oct '09

06 Oct '09

I just set up a new system running an Asus P7P55D PRO motherboard. It looks like this uses the VIA VT1828S codec. The analog output seems to work but the SPDIF output isn't recognized. Originally tried 2.6.30.5, but Linus' current git doesn't seem any different. The device only gets detected as HDA Generic, which, from my reading of the code, ignores SPDIF outputs. I tried hacking patch_via.c to add the codec ID to match existing types like VT1708S and VT1702, each of those got the SPDIF output to show up, but no output showed up on the receiver, and also the analog output then stopped being recognized (looks like some node IDs don't match). Is there some extra support that needs to be added for this codec, or anything else I can try to get this working? The SPDIF does work in Windows 7 with just the standard Microsoft HDA driver.. aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: HDA Generic [HDA Generic] Subdevices: 1/1 Subdevice #0: subdevice #0 The codec data seems to have references to the SPDIF output, but no device is actually getting created. The data is below. Any suggestions? cat /proc/asound/card0/codec#0 Codec: VIA ID 4441 Address: 0 Function Id: 0x1 Vendor Id: 0x11064441 Subsystem Id: 0x104383cf Revision Id: 0x100100 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=1, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x08 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x2a 0x2a] Converter: stream=5, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x09 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x26 0x26] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x0a [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x26 0x26] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x0b [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x26 0x26] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x0c [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x2a 0x2a] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x0d [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0e [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x0f [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x10 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x14 0x14] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x1e Node 0x11 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x8b 0x8b] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x1f Node 0x12 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x15 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x16 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x18 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Power: setting=D0, actual=D0 Connection: 2 0x08 0x21 Node 0x19 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Power: setting=D0, actual=D0 Connection: 2 0x09 0x21 Node 0x1a [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Power: setting=D0, actual=D0 Connection: 2 0x0b 0x21 Node 0x1b [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Power: setting=D0, actual=D0 Connection: 2 0x34 0x21 Node 0x1c [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Power: setting=D0, actual=D0 Connection: 2 0x35 0x21 Node 0x1d [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1e [Audio Selector] wcaps 0x300501: Stereo Power: setting=D0, actual=D0 Connection: 6 0x2c* 0x2b 0x2a 0x29 0x28 0x21 Node 0x1f [Audio Selector] wcaps 0x300501: Stereo Power: setting=D0, actual=D0 Connection: 6 0x2c 0x2b* 0x2a 0x29 0x28 0x21 Node 0x20 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x21 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] Power: setting=D0, actual=D0 Connection: 5 0x2c 0x2b 0x2a 0x29 0x28 Node 0x22 [Beep Generator Widget] wcaps 0x70040c: Mono Amp-Out Amp-Out caps: ofs=0x0a, nsteps=0x12, stepsize=0x05, mute=1 Amp-Out vals: [0x00] Power: setting=D0, actual=D0 Node 0x23 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x24 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001001c: OUT HP EAPD Detect EAPD 0x0: Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x18 Node 0x25 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000001c: OUT HP Detect Pin Default 0x01011012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x19 Node 0x26 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00000014: OUT Detect Pin Default 0x01016011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x0a Node 0x27 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00000014: OUT Detect Pin Default 0x01012014: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x1a Node 0x28 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x0221401f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x1b Node 0x29 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x02a19037: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x7 Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x1c Node 0x2a [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00002334: IN OUT Detect Vref caps: HIZ 50 100 Pin Default 0x0181303e: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x3, Sequence = 0xe Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 2 0x09* 0x0c Node 0x2b [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00002334: IN OUT Detect Vref caps: HIZ 50 100 Pin Default 0x01a19036: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x6 Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 2 0x0a* 0x0c Node 0x2c [Pin Complex] wcaps 0x400401: Stereo Pincap 0x00000020: IN Pin Default 0x90331138: [Fixed] CD at Int N/A Conn = ATAPI, Color = Black DefAssociation = 0x3, Sequence = 0x8 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Power: setting=D0, actual=D0 Node 0x2d [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x00000090: OUT HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Power: setting=D0, actual=D0 Connection: 1 0x0e Node 0x2e [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x074411f0: [Jack] SPDIF Out at Ext Rear Panel Conn = RCA, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power: setting=D0, actual=D0 Connection: 1 0x0f Node 0x2f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x30 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x31 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x32 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x33 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x34 [Audio Selector] wcaps 0x300501: Stereo Power: setting=D0, actual=D0 Connection: 3 0x08* 0x0b 0x0c Node 0x35 [Audio Selector] wcaps 0x300501: Stereo Power: setting=D0, actual=D0 Connection: 3 0x08* 0x0b 0x0c

4 10

[alsa-devel] [PATCH 1/7] ak4114 - fix errors in output selector bits
by pavel.hofman＠ivitera.com 06 Oct '09

06 Oct '09

From: Pavel Hofman <pavel.hofman(a)ivitera.com> * the previous version had a typo - values of AK4114_OPS10-12 were identical with AK4114_OPS00-02 * Since no cards actually use this feature, the bug was not identified earlier Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> diff --git a/include/ak4114.h b/include/ak4114.h index d293d36..3ce69fd 100644 --- a/include/ak4114.h +++ b/include/ak4114.h @@ -95,13 +95,13 @@ /* AK4114_REG_IO0 */ #define AK4114_TX1E (1<<7) /* TX1 Output Enable (1 = enable) */ -#define AK4114_OPS12 (1<<2) /* Output Though Data Selector for TX1 pin */ -#define AK4114_OPS11 (1<<1) /* Output Though Data Selector for TX1 pin */ -#define AK4114_OPS10 (1<<0) /* Output Though Data Selector for TX1 pin */ +#define AK4114_OPS12 (1<<6) /* Output Data Selector for TX1 pin */ +#define AK4114_OPS11 (1<<5) /* Output Data Selector for TX1 pin */ +#define AK4114_OPS10 (1<<4) /* Output Data Selector for TX1 pin */ #define AK4114_TX0E (1<<3) /* TX0 Output Enable (1 = enable) */ -#define AK4114_OPS02 (1<<2) /* Output Though Data Selector for TX0 pin */ -#define AK4114_OPS01 (1<<1) /* Output Though Data Selector for TX0 pin */ -#define AK4114_OPS00 (1<<0) /* Output Though Data Selector for TX0 pin */ +#define AK4114_OPS02 (1<<2) /* Output Data Selector for TX0 pin */ +#define AK4114_OPS01 (1<<1) /* Output Data Selector for TX0 pin */ +#define AK4114_OPS00 (1<<0) /* Output Data Selector for TX0 pin */ /* AK4114_REG_IO1 */ #define AK4114_EFH1 (1<<7) /* Interrupt 0 pin Hold */ -- 1.5.6.3

3 17

[alsa-devel] [PATCH] sound: Don't assume i2c device probing always succeeds
by Jean Delvare 05 Oct '09

05 Oct '09

If i2c device probing fails, then there is no driver to dereference after calling i2c_new_device(). Stop assuming that probing will always succeed, to avoid NULL pointer dereferences. We have an easier access to the driver anyway. Reported-by: Tim Shepard <shep(a)alum.mit.edu> Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Johannes Berg <johannes(a)sipsolutions.net> --- The code is similar to the one in therm_adt746x, for which Tim reported a real-world oops, so it should be fixed ASAP. sound/aoa/codecs/onyx.c | 4 +++- sound/aoa/codecs/tas.c | 4 +++- sound/ppc/keywest.c | 4 +++- 3 files changed, 9 insertions(+), 3 deletions(-) --- linux-2.6.32-rc1.orig/sound/aoa/codecs/onyx.c 2009-09-30 15:13:12.000000000 +0200 +++ linux-2.6.32-rc1/sound/aoa/codecs/onyx.c 2009-09-30 15:13:58.000000000 +0200 @@ -996,6 +996,8 @@ static void onyx_exit_codec(struct aoa_c onyx->codec.soundbus_dev->detach_codec(onyx->codec.soundbus_dev, onyx); } +static struct i2c_driver onyx_driver; + static int onyx_create(struct i2c_adapter *adapter, struct device_node *node, int addr) @@ -1027,7 +1029,7 @@ static int onyx_create(struct i2c_adapte * Let i2c-core delete that device on driver removal. * This is safe because i2c-core holds the core_lock mutex for us. */ - list_add_tail(&client->detected, &client->driver->clients); + list_add_tail(&client->detected, &onyx_driver.clients); return 0; } --- linux-2.6.32-rc1.orig/sound/aoa/codecs/tas.c 2009-09-30 15:13:12.000000000 +0200 +++ linux-2.6.32-rc1/sound/aoa/codecs/tas.c 2009-09-30 15:13:58.000000000 +0200 @@ -882,6 +882,8 @@ static void tas_exit_codec(struct aoa_co } +static struct i2c_driver tas_driver; + static int tas_create(struct i2c_adapter *adapter, struct device_node *node, int addr) @@ -902,7 +904,7 @@ static int tas_create(struct i2c_adapter * Let i2c-core delete that device on driver removal. * This is safe because i2c-core holds the core_lock mutex for us. */ - list_add_tail(&client->detected, &client->driver->clients); + list_add_tail(&client->detected, &tas_driver.clients); return 0; } --- linux-2.6.32-rc1.orig/sound/ppc/keywest.c 2009-09-30 15:13:12.000000000 +0200 +++ linux-2.6.32-rc1/sound/ppc/keywest.c 2009-09-30 15:13:58.000000000 +0200 @@ -40,6 +40,8 @@ static int keywest_probe(struct i2c_clie return 0; } +struct i2c_driver keywest_driver; + /* * This is kind of a hack, best would be to turn powermac to fixed i2c * bus numbers and declare the sound device as part of platform @@ -65,7 +67,7 @@ static int keywest_attach_adapter(struct * This is safe because i2c-core holds the core_lock mutex for us. */ list_add_tail(&keywest_ctx->client->detected, - &keywest_ctx->client->driver->clients); + &keywest_driver.clients); return 0; } -- Jean Delvare

3 9

Re: [alsa-devel] My Codec: IDT 92HD75B3X5 issues
by Munzir Taha (منذر طه) 02 Oct '09

02 Oct '09

On Yau al-Thulatha 09 Shawwal 1430 5:37:04 pm you wrote: > At Tue, 29 Sep 2009 16:24:29 +0300, > > Munzir Taha (=?utf-8?q?=D9=85=D9=86=D8=B0=D8=B1?= =?utf-8?q?_=D8=B7=D9=87?=) wrote: > > On Yau al-Thulatha 09 Shawwal 1430 9:24:19 am Takashi Iwai wrote: > > > At Tue, 29 Sep 2009 07:40:56 +0300, > > > > > > Munzir Taha (=?utf-8?q?=D9=85=D9=86=D8=B0=D8=B1?= > > > > =?utf-8?q?_=D8=B7=D9=87?=) wrote: > > > > On Yaum al-Ithnain 08 Shawwal 1430 11:41:52 am Takashi Iwai wrote: > > > > > At Mon, 28 Sep 2009 06:58:46 +0300, > > > > > > > > > > Munzir Taha (=?utf-8?q?=D9=85=D9=86=D8=B0=D8=B1?= > > > > > > > > =?utf-8?q?_=D8=B7=D9=87?=) wrote: > > > > > > Hi sirs, > > > > > > These are the issues I have with my Laptop HP Pavilion dv6 1225ee > > > > > > regarding audio support. > > > > > > > > > > > > 1. No sense for headphone jack (jack_present?), when headphone is > > > > > > plugged (present_int_hp?) > > > > > > > > > > > > 2. There headphone mic jack is not working. The one next to the > > > > > > webcam is always recording even when I change the "Mic Jack" > > > > > > control from "Mic In" to "Line In" > > > > > > > > > > Make sure that you use alsa-driver-snapshot tarball instead of > > > > > 1.0.21 released version. There are many fixes since 1.0.21 > > > > > release. > > > > > > > > > > > > > > > > > > > > ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-dr > > > > >iver -sn apshot.tar.gz > > > > > > > > Thanks a lot Takashi, things are better now. However, when I tested > > > > the internal mic it's very noisy. I played a bit with the capture and > > > > digital controls but still compared with Windows, it's lame. > > > > > > OK, could you give alsa-info.sh output at this state? > > > Please run with --no-upload option, and attach the generated file. > > > > Attached. > > Try to run: > > % amixer -c0 set Capture cap 12dB > % amixer -c0 set "Digital Mic" 10dB > % amixer -c0 set Digital 0dB > > Any difference? No, still there is a clear shshshshshsh sound in the background. Please check the attachment for a short arecord -f cd without me speaking on the mic and still you can hear this noise very clear. Also attached is the alsa-info report at the time of recording -- Munzir Taha Telecommunications and Electronics Engineer Master CIW Designer, ICDL, MOUS, Linux+, LPI, SCSA Riyadh, SA

2 13

[alsa-devel] [PATCH] Don't check if period size is equal to buffer size
by Lubomir Rintel 02 Oct '09

02 Oct '09

At least some drivers seem to allow a single period and those which don't would correct us anyway. Moreover, Pulseaudio with tsched sets this frequently. --- aplay/aplay.c | 5 ----- 1 files changed, 0 insertions(+), 5 deletions(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index c7c82a1..4fa36cb 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -1044,11 +1044,6 @@ static void set_params(void) } snd_pcm_hw_params_get_period_size(params, &chunk_size, 0); snd_pcm_hw_params_get_buffer_size(params, &buffer_size); - if (chunk_size == buffer_size) { - error(_("Can't use period equal to buffer size (%lu == %lu)"), - chunk_size, buffer_size); - exit(EXIT_FAILURE); - } snd_pcm_sw_params_current(handle, swparams); if (avail_min < 0) n = chunk_size; -- 1.6.2.5

3 2

[alsa-devel] Patches for alsactl init db
by Dan Chen 02 Oct '09

02 Oct '09

Hi, Attached are two patches against alsa-utils.git HEAD that create an intel8x0 init db file for ICH4. This is necessary from bug reports in Debian and Ubuntu that state that External Amplifier must be muted to have audible (analog) playback by default on the Sony VAIO VGN-T350P. Thanks, Dan

3 5

[alsa-devel] [PATCH 1/3] ASoC: DaVinci: i2s, reduce underruns by combining into 1 element
by Troy Kisky 01 Oct '09

01 Oct '09

Allow the left and right 16 bit samples to be shifted out as 1 32 bit sample. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> -- This applies to Kevin's temp/asoc branch --- arch/arm/mach-davinci/include/mach/asp.h | 6 ++ sound/soc/davinci/davinci-i2s.c | 74 ++++++++++++++++++++++-------- 2 files changed, 61 insertions(+), 19 deletions(-) diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 18e4ce3..a3d2aa1 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -51,6 +51,12 @@ struct snd_platform_data { u32 rx_dma_offset; enum dma_event_q eventq_no; /* event queue number */ unsigned int codec_fmt; + /* + * Allowing this is more efficient and eliminates left and right swaps + * caused by underruns, but will swap the left and right channels + * when compared to previous behavior. + */ + unsigned disable_channel_combine:1; /* McASP specific fields */ int tdm_slots; diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 12a6c54..081b2d4 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -97,6 +97,23 @@ enum { DAVINCI_MCBSP_WORD_32, }; +static const unsigned char data_type[SNDRV_PCM_FORMAT_S32_LE + 1] = { + [SNDRV_PCM_FORMAT_S8] = 1, + [SNDRV_PCM_FORMAT_S16_LE] = 2, + [SNDRV_PCM_FORMAT_S32_LE] = 4, +}; + +static const unsigned char asp_word_length[SNDRV_PCM_FORMAT_S32_LE + 1] = { + [SNDRV_PCM_FORMAT_S8] = DAVINCI_MCBSP_WORD_8, + [SNDRV_PCM_FORMAT_S16_LE] = DAVINCI_MCBSP_WORD_16, + [SNDRV_PCM_FORMAT_S32_LE] = DAVINCI_MCBSP_WORD_32, +}; + +static const unsigned char double_fmt[SNDRV_PCM_FORMAT_S32_LE + 1] = { + [SNDRV_PCM_FORMAT_S8] = SNDRV_PCM_FORMAT_S16_LE, + [SNDRV_PCM_FORMAT_S16_LE] = SNDRV_PCM_FORMAT_S32_LE, +}; + static struct davinci_pcm_dma_params davinci_i2s_pcm_out = { .name = "I2S PCM Stereo out", }; @@ -113,6 +130,27 @@ struct davinci_mcbsp_dev { u32 pcr; struct clk *clk; struct davinci_pcm_dma_params *dma_params[2]; + /* + * Combining both channels into 1 element will at least double the + * amount of time between servicing the dma channel, increase + * effiency, and reduce the chance of overrun/underrun. But, + * it will result in the left & right channels being swapped. + * + * If relabeling the left and right channels is not possible, + * you may want to let the codec know to swap them back. + * + * It may allow x10 the amount of time to service dma requests, + * if the codec is master and is using an unnecessarily fast bit clock + * (ie. tlvaic23b), independent of the sample rate. So, having an + * entire frame at once means it can be serviced at the sample rate + * instead of the bit clock rate. + * + * In the now unlikely case that an underrun still + * occurs, both the left and right samples will be repeated + * so that no pops are heard, and the left and right channels + * won't end up being swapped because of the underrun. + */ + unsigned disable_channel_combine:1; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -359,6 +397,8 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, int mcbsp_word_length; unsigned int rcr, xcr, srgr; u32 spcr; + snd_pcm_format_t fmt; + unsigned element_cnt = 1; /* general line settings */ spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); @@ -388,27 +428,22 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, xcr |= DAVINCI_MCBSP_XCR_XDATDLY(1); } /* Determine xfer data type */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - dma_params->data_type = 1; - mcbsp_word_length = DAVINCI_MCBSP_WORD_8; - break; - case SNDRV_PCM_FORMAT_S16_LE: - dma_params->data_type = 2; - mcbsp_word_length = DAVINCI_MCBSP_WORD_16; - break; - case SNDRV_PCM_FORMAT_S32_LE: - dma_params->data_type = 4; - mcbsp_word_length = DAVINCI_MCBSP_WORD_32; - break; - default: + fmt = params_format(params); + if ((fmt > SNDRV_PCM_FORMAT_S32_LE) || !data_type[fmt]) { printk(KERN_WARNING "davinci-i2s: unsupported PCM format\n"); return -EINVAL; } - - dma_params->acnt = dma_params->data_type; - rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(1); - xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(1); + if (params_channels(params) == 2) { + element_cnt = 2; + if (double_fmt[fmt] && !dev->disable_channel_combine) { + element_cnt = 1; + fmt = double_fmt[fmt]; + } + } + dma_params->acnt = dma_params->data_type = data_type[fmt]; + mcbsp_word_length = asp_word_length[fmt]; + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); rcr |= DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length); @@ -524,7 +559,8 @@ static int davinci_i2s_probe(struct platform_device *pdev) ret = -ENOMEM; goto err_release_region; } - + if (pdata) + dev->disable_channel_combine = pdata->disable_channel_combine; dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { ret = -ENODEV; -- 1.5.6.3

6 28

[alsa-devel] [PATCH 0/1] ASoC: debugfs support improvement
by Peter Ujfalusi 01 Oct '09

01 Oct '09

Hello, I have noticed that the support for multiple cards/codecs via the debugfs is kind of broken. The second codec, which tries to create the debugfs files would emit the following messages: ASoC: Failed to create codec register debugfs file Failed to create pop time debugfs file ... The following patch fixes this by adding a directory per codec to host the files. The additional directory is named as codec->name, so it is easier to find the correct codec we are looking for. --- Peter Ujfalusi (1): ASoC: add support for multiple cards/codecs in debugfs include/sound/soc.h | 1 + sound/soc/soc-core.c | 21 ++++++++++++++------- 2 files changed, 15 insertions(+), 7 deletions(-)

2 6

[alsa-devel] [PATCH] sscape: convert to firmware loader framework
by Krzysztof Helt 01 Oct '09

01 Oct '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The conversion solves the problem that firmware size was set to 64KB while non PnP cards have 128KB firmware files. An additional firmware initialization code has been moved from the OSS driver. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This is the second version with fixes to issues pointed by Takashi and two mistakes in the original patch. Documentation/sound/alsa/ALSA-Configuration.txt | 8 +- include/sound/sscape_ioctl.h | 21 -- sound/isa/Kconfig | 8 +- sound/isa/sscape.c | 328 ++++++++--------------- 4 files changed, 116 insertions(+), 249 deletions(-) delete mode 100644 include/sound/sscape_ioctl.h diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 1c8eb45..cf98525 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1631,7 +1631,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module snd-sscape ----------------- - Module for ENSONIQ SoundScape PnP cards. + Module for ENSONIQ SoundScape cards. port - Port # (PnP setup) wss_port - WSS Port # (PnP setup) @@ -1640,9 +1640,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. dma - DMA # (PnP setup) dma2 - 2nd DMA # (PnP setup, -1 to disable) - This module supports multiple cards. ISA PnP must be enabled. - You need sscape_ctl tool in alsa-tools package for loading - the microcode. + This module supports multiple cards. + + The driver requires the firmware loader support on kernel. Module snd-sun-amd7930 (on sparc only) -------------------------------------- diff --git a/include/sound/sscape_ioctl.h b/include/sound/sscape_ioctl.h deleted file mode 100644 index 0d88859..0000000 --- a/include/sound/sscape_ioctl.h +++ /dev/null @@ -1,21 +0,0 @@ -#ifndef SSCAPE_IOCTL_H -#define SSCAPE_IOCTL_H - - -struct sscape_bootblock -{ - unsigned char code[256]; - unsigned version; -}; - -#define SSCAPE_MICROCODE_SIZE 65536 - -struct sscape_microcode -{ - unsigned char __user *code; -}; - -#define SND_SSCAPE_LOAD_BOOTB _IOWR('P', 100, struct sscape_bootblock) -#define SND_SSCAPE_LOAD_MCODE _IOW ('P', 101, struct sscape_microcode) - -#endif diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index b90fc16..02fe81c 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -372,9 +372,9 @@ config SND_SGALAXY config SND_SSCAPE tristate "Ensoniq SoundScape driver" - select SND_HWDEP select SND_MPU401_UART select SND_WSS_LIB + select FW_LOADER help Say Y here to include support for Ensoniq SoundScape and Ensoniq OEM soundcards. @@ -382,7 +382,11 @@ config SND_SSCAPE The PCM audio is supported on SoundScape Classic, Elite, PnP and VIVO cards. The supported OEM cards are SPEA Media FX and Reveal SC-600. - The MIDI support is very experimental. + The MIDI support is very experimental and requires binary + firmware files called "scope.cod" and "sndscape.co?" where the + ? is digit 0, 1, 2, 3 or 4. The firmware files can be found + in DOS or Windows driver packages. One has to put the firmware + files into the /lib/firmware directory. To compile this driver as a module, choose M here: the module will be called snd-sscape. diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index b11c35f..1ce465c 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -1,5 +1,5 @@ /* - * Low-level ALSA driver for the ENSONIQ SoundScape PnP + * Low-level ALSA driver for the ENSONIQ SoundScape * Copyright (c) by Chris Rankin * * This driver was written in part using information obtained from @@ -25,22 +25,26 @@ #include <linux/err.h> #include <linux/isa.h> #include <linux/delay.h> +#include <linux/firmware.h> #include <linux/pnp.h> #include <linux/spinlock.h> #include <linux/moduleparam.h> #include <asm/dma.h> #include <sound/core.h> -#include <sound/hwdep.h> #include <sound/wss.h> #include <sound/mpu401.h> #include <sound/initval.h> -#include <sound/sscape_ioctl.h> - MODULE_AUTHOR("Chris Rankin"); -MODULE_DESCRIPTION("ENSONIQ SoundScape PnP driver"); +MODULE_DESCRIPTION("ENSONIQ SoundScape driver"); MODULE_LICENSE("GPL"); +MODULE_FIRMWARE("sndscape.co0"); +MODULE_FIRMWARE("sndscape.co1"); +MODULE_FIRMWARE("sndscape.co2"); +MODULE_FIRMWARE("sndscape.co3"); +MODULE_FIRMWARE("sndscape.co4"); +MODULE_FIRMWARE("scope.cod"); static int index[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IDX; static char* id[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_STR; @@ -142,14 +146,12 @@ struct soundscape { struct resource *wss_res; struct snd_wss *chip; struct snd_mpu401 *mpu; - struct snd_hwdep *hw; /* * The MIDI device won't work until we've loaded * its firmware via a hardware-dependent device IOCTL */ spinlock_t fwlock; - int hw_in_use; unsigned long midi_usage; unsigned char midi_vol; }; @@ -167,12 +169,6 @@ static inline struct soundscape *get_mpu401_soundscape(struct snd_mpu401 * mpu) return (struct soundscape *) (mpu->private_data); } -static inline struct soundscape *get_hwdep_soundscape(struct snd_hwdep * hw) -{ - return (struct soundscape *) (hw->private_data); -} - - /* * Allocates some kernel memory that we can use for DMA. * I think this means that the memory has to map to @@ -393,12 +389,12 @@ static int obp_startup_ack(struct soundscape *s, unsigned timeout) do { unsigned long flags; - unsigned char x; + int x; spin_lock_irqsave(&s->lock, flags); - x = inb(HOST_DATA_IO(s->io_base)); + x = host_read_unsafe(s->io_base); spin_unlock_irqrestore(&s->lock, flags); - if ((x & 0xfe) == 0xfe) + if (x == 0xfe || x == 0xff) return 1; msleep(10); @@ -420,10 +416,10 @@ static int host_startup_ack(struct soundscape *s, unsigned timeout) do { unsigned long flags; - unsigned char x; + int x; spin_lock_irqsave(&s->lock, flags); - x = inb(HOST_DATA_IO(s->io_base)); + x = host_read_unsafe(s->io_base); spin_unlock_irqrestore(&s->lock, flags); if (x == 0xfe) return 1; @@ -438,14 +434,14 @@ static int host_startup_ack(struct soundscape *s, unsigned timeout) * Upload a byte-stream into the SoundScape using DMA channel A. */ static int upload_dma_data(struct soundscape *s, - const unsigned char __user *data, + const unsigned char *data, size_t size) { unsigned long flags; struct snd_dma_buffer dma; int ret; - if (!get_dmabuf(&dma, PAGE_ALIGN(size))) + if (!get_dmabuf(&dma, PAGE_ALIGN(32 * 1024))) return -ENOMEM; spin_lock_irqsave(&s->lock, flags); @@ -458,7 +454,6 @@ static int upload_dma_data(struct soundscape *s, /* * Enable the DMA channels and configure them ... */ - sscape_write_unsafe(s->io_base, GA_DMACFG_REG, 0x50); sscape_write_unsafe(s->io_base, GA_DMAA_REG, (s->chip->dma1 << 4) | DMA_8BIT); sscape_write_unsafe(s->io_base, GA_DMAB_REG, 0x20); @@ -468,35 +463,17 @@ static int upload_dma_data(struct soundscape *s, sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) | 0x80); /* - * Upload the user's data (firmware?) to the SoundScape + * Upload the firmware to the SoundScape * board through the DMA channel ... */ while (size != 0) { unsigned long len; - /* - * Apparently, copying to/from userspace can sleep. - * We are therefore forbidden from holding any - * spinlocks while we copy ... - */ - spin_unlock_irqrestore(&s->lock, flags); - - /* - * Remember that the data that we want to DMA - * comes from USERSPACE. We have already verified - * the userspace pointer ... - */ len = min(size, dma.bytes); - len -= __copy_from_user(dma.area, data, len); + memcpy(dma.area, data, len); data += len; size -= len; - /* - * Grab that spinlock again, now that we've - * finished copying! - */ - spin_lock_irqsave(&s->lock, flags); - snd_dma_program(s->chip->dma1, dma.addr, len, DMA_MODE_WRITE); sscape_start_dma_unsafe(s->io_base, GA_DMAA_REG); if (!sscape_wait_dma_unsafe(s->io_base, GA_DMAA_REG, 5000)) { @@ -512,6 +489,7 @@ static int upload_dma_data(struct soundscape *s, } /* while */ set_host_mode_unsafe(s->io_base); + outb(0x0, s->io_base); /* * Boot the board ... (I think) @@ -537,7 +515,7 @@ _release_dma: /* * NOTE!!! We are NOT holding any spinlocks at this point !!! */ - sscape_write(s, GA_DMAA_REG, (s->ic_type == IC_ODIE ? 0x70 : 0x40)); + sscape_write(s, GA_DMAA_REG, (s->ic_type == IC_OPUS ? 0x40 : 0x70)); free_dmabuf(&dma); return ret; @@ -547,162 +525,69 @@ _release_dma: * Upload the bootblock(?) into the SoundScape. The only * purpose of this block of code seems to be to tell * us which version of the microcode we should be using. - * - * NOTE: The boot-block data resides in USER-SPACE!!! - * However, we have already verified its memory - * addresses by the time we get here. */ -static int sscape_upload_bootblock(struct soundscape *sscape, struct sscape_bootblock __user *bb) +static int sscape_upload_bootblock(struct snd_card *card) { + struct soundscape *sscape = get_card_soundscape(card); unsigned long flags; + const struct firmware *init_fw = NULL; int data = 0; int ret; - ret = upload_dma_data(sscape, bb->code, sizeof(bb->code)); - - spin_lock_irqsave(&sscape->lock, flags); - if (ret == 0) { - data = host_read_ctrl_unsafe(sscape->io_base, 100); - } - set_midi_mode_unsafe(sscape->io_base); - spin_unlock_irqrestore(&sscape->lock, flags); - - if (ret == 0) { - if (data < 0) { - snd_printk(KERN_ERR "sscape: timeout reading firmware version\n"); - ret = -EAGAIN; - } - else if (__copy_to_user(&bb->version, &data, sizeof(bb->version))) { - ret = -EFAULT; - } + ret = request_firmware(&init_fw, "scope.cod", card->dev); + if (ret < 0) { + snd_printk(KERN_ERR "Error loading scope.cod"); + return ret; } + ret = upload_dma_data(sscape, init_fw->data, init_fw->size); - return ret; -} + release_firmware(init_fw); -/* - * Upload the microcode into the SoundScape. The - * microcode is 64K of data, and if we try to copy - * it into a local variable then we will SMASH THE - * KERNEL'S STACK! We therefore leave it in USER - * SPACE, and save ourselves from copying it at all. - */ -static int sscape_upload_microcode(struct soundscape *sscape, - const struct sscape_microcode __user *mc) -{ - unsigned long flags; - char __user *code; - int err; - - /* - * We are going to have to copy this data into a special - * DMA-able buffer before we can upload it. We shall therefore - * just check that the data pointer is valid for now. - * - * NOTE: This buffer is 64K long! That's WAY too big to - * copy into a stack-temporary anyway. - */ - if ( get_user(code, &mc->code) || - !access_ok(VERIFY_READ, code, SSCAPE_MICROCODE_SIZE) ) - return -EFAULT; + spin_lock_irqsave(&sscape->lock, flags); + if (ret == 0) + data = host_read_ctrl_unsafe(sscape->io_base, 100); - if ((err = upload_dma_data(sscape, code, SSCAPE_MICROCODE_SIZE)) == 0) { - snd_printk(KERN_INFO "sscape: MIDI firmware loaded\n"); - } + if (data & 0x10) + sscape_write_unsafe(sscape->io_base, GA_SMCFGA_REG, 0x2f); - spin_lock_irqsave(&sscape->lock, flags); - set_midi_mode_unsafe(sscape->io_base); spin_unlock_irqrestore(&sscape->lock, flags); - initialise_mpu401(sscape->mpu); + data &= 0xf; + if (ret == 0 && data > 7) { + snd_printk(KERN_ERR "timeout reading firmware version\n"); + ret = -EAGAIN; + } - return err; + return (ret == 0) ? data : ret; } /* - * Hardware-specific device functions, to implement special - * IOCTLs for the SoundScape card. This is how we upload - * the microcode into the card, for example, and so we - * must ensure that no two processes can open this device - * simultaneously, and that we can't open it at all if - * someone is using the MIDI device. + * Upload the microcode into the SoundScape. */ -static int sscape_hw_open(struct snd_hwdep * hw, struct file *file) +static int sscape_upload_microcode(struct snd_card *card, int version) { - register struct soundscape *sscape = get_hwdep_soundscape(hw); - unsigned long flags; + struct soundscape *sscape = get_card_soundscape(card); + const struct firmware *init_fw = NULL; + char name[14]; int err; - spin_lock_irqsave(&sscape->fwlock, flags); + snprintf(name, sizeof(name), "sndscape.co%d", version); - if ((sscape->midi_usage != 0) || sscape->hw_in_use) { - err = -EBUSY; - } else { - sscape->hw_in_use = 1; - err = 0; + err = request_firmware(&init_fw, name, card->dev); + if (err < 0) { + snd_printk(KERN_ERR "Error loading sndscape.co%d", version); + return err; } + err = upload_dma_data(sscape, init_fw->data, init_fw->size); + if (err == 0) + snd_printk(KERN_INFO "MIDI firmware loaded %d KBs\n", + init_fw->size >> 10); - spin_unlock_irqrestore(&sscape->fwlock, flags); - return err; -} - -static int sscape_hw_release(struct snd_hwdep * hw, struct file *file) -{ - register struct soundscape *sscape = get_hwdep_soundscape(hw); - unsigned long flags; - - spin_lock_irqsave(&sscape->fwlock, flags); - sscape->hw_in_use = 0; - spin_unlock_irqrestore(&sscape->fwlock, flags); - return 0; -} - -static int sscape_hw_ioctl(struct snd_hwdep * hw, struct file *file, - unsigned int cmd, unsigned long arg) -{ - struct soundscape *sscape = get_hwdep_soundscape(hw); - int err = -EBUSY; - - switch (cmd) { - case SND_SSCAPE_LOAD_BOOTB: - { - register struct sscape_bootblock __user *bb = (struct sscape_bootblock __user *) arg; - - /* - * We are going to have to copy this data into a special - * DMA-able buffer before we can upload it. We shall therefore - * just check that the data pointer is valid for now ... - */ - if ( !access_ok(VERIFY_READ, bb->code, sizeof(bb->code)) ) - return -EFAULT; - - /* - * Now check that we can write the firmware version number too... - */ - if ( !access_ok(VERIFY_WRITE, &bb->version, sizeof(bb->version)) ) - return -EFAULT; - - err = sscape_upload_bootblock(sscape, bb); - } - break; - - case SND_SSCAPE_LOAD_MCODE: - { - register const struct sscape_microcode __user *mc = (const struct sscape_microcode __user *) arg; - - err = sscape_upload_microcode(sscape, mc); - } - break; - - default: - err = -EINVAL; - break; - } /* switch */ + release_firmware(init_fw); return err; } - /* * Mixer control for the SoundScape's MIDI device. */ @@ -920,7 +805,7 @@ static int mpu401_open(struct snd_mpu401 * mpu) spin_lock_irqsave(&sscape->fwlock, flags); - if (sscape->hw_in_use || (sscape->midi_usage == ULONG_MAX)) { + if (sscape->midi_usage == ULONG_MAX) { err = -EBUSY; } else { ++(sscape->midi_usage); @@ -1053,13 +938,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, } } - strcpy(card->driver, "SoundScape"); - strcpy(card->shortname, pcm->name); - snprintf(card->longname, sizeof(card->longname), - "%s at 0x%lx, IRQ %d, DMA1 %d, DMA2 %d\n", - pcm->name, chip->port, chip->irq, - chip->dma1, chip->dma2); - sscape->chip = chip; } @@ -1162,29 +1040,6 @@ static int __devinit create_sscape(int dev, struct snd_card *card) return -ENXIO; } - if (sscape->type != SSCAPE_VIVO) { - /* - * Now create the hardware-specific device so that we can - * load the microcode into the on-board processor. - * We cannot use the MPU-401 MIDI system until this firmware - * has been loaded into the card. - */ - err = snd_hwdep_new(card, "MC68EC000", 0, &(sscape->hw)); - if (err < 0) { - printk(KERN_ERR "sscape: Failed to create " - "firmware device\n"); - goto _release_dma; - } - strlcpy(sscape->hw->name, "SoundScape M68K", - sizeof(sscape->hw->name)); - sscape->hw->name[sizeof(sscape->hw->name) - 1] = '\0'; - sscape->hw->iface = SNDRV_HWDEP_IFACE_SSCAPE; - sscape->hw->ops.open = sscape_hw_open; - sscape->hw->ops.release = sscape_hw_release; - sscape->hw->ops.ioctl = sscape_hw_ioctl; - sscape->hw->private_data = sscape; - } - /* * Tell the on-board devices where their resources are (I think - * I can't be sure without a datasheet ... So many magic values!) @@ -1222,28 +1077,56 @@ static int __devinit create_sscape(int dev, struct snd_card *card) wss_port[dev], irq[dev]); goto _release_dma; } + strcpy(card->driver, "SoundScape"); + strcpy(card->shortname, name); + snprintf(card->longname, sizeof(card->longname), + "%s at 0x%lx, IRQ %d, DMA1 %d, DMA2 %d\n", + name, sscape->chip->port, sscape->chip->irq, + sscape->chip->dma1, sscape->chip->dma2); + #define MIDI_DEVNUM 0 if (sscape->type != SSCAPE_VIVO) { - err = create_mpu401(card, MIDI_DEVNUM, port[dev], mpu_irq[dev]); - if (err < 0) { - printk(KERN_ERR "sscape: Failed to create " - "MPU-401 device at 0x%lx\n", - port[dev]); - goto _release_dma; - } + err = sscape_upload_bootblock(card); + if (err >= 0) + err = sscape_upload_microcode(card, err); - /* - * Enable the master IRQ ... - */ - sscape_write(sscape, GA_INTENA_REG, 0x80); - - /* - * Initialize mixer - */ - sscape->midi_vol = 0; - host_write_ctrl_unsafe(sscape->io_base, CMD_SET_MIDI_VOL, 100); - host_write_ctrl_unsafe(sscape->io_base, 0, 100); - host_write_ctrl_unsafe(sscape->io_base, CMD_XXX_MIDI_VOL, 100); + if (err == 0) { + err = create_mpu401(card, MIDI_DEVNUM, port[dev], + mpu_irq[dev]); + if (err < 0) { + printk(KERN_ERR "sscape: Failed to create " + "MPU-401 device at 0x%lx\n", + port[dev]); + goto _release_dma; + } + + /* + * Enable the master IRQ ... + */ + sscape_write(sscape, GA_INTENA_REG, 0x80); + + /* + * Initialize mixer + */ + spin_lock_irqsave(&sscape->lock, flags); + sscape->midi_vol = 0; + host_write_ctrl_unsafe(sscape->io_base, + CMD_SET_MIDI_VOL, 100); + host_write_ctrl_unsafe(sscape->io_base, + sscape->midi_vol, 100); + host_write_ctrl_unsafe(sscape->io_base, + CMD_XXX_MIDI_VOL, 100); + host_write_ctrl_unsafe(sscape->io_base, + sscape->midi_vol, 100); + host_write_ctrl_unsafe(sscape->io_base, + CMD_SET_EXTMIDI, 100); + host_write_ctrl_unsafe(sscape->io_base, + 0, 100); + host_write_ctrl_unsafe(sscape->io_base, CMD_ACK, 100); + + set_midi_mode_unsafe(sscape->io_base); + spin_unlock_irqrestore(&sscape->lock, flags); + } } /* @@ -1301,11 +1184,12 @@ static int __devinit snd_sscape_probe(struct device *pdev, unsigned int dev) sscape->type = SSCAPE; dma[dev] &= 0x03; + snd_card_set_dev(card, pdev); + ret = create_sscape(dev, card); if (ret < 0) goto _release_card; - snd_card_set_dev(card, pdev); if ((ret = snd_card_register(card)) < 0) { printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; @@ -1426,12 +1310,12 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, wss_port[idx] = pnp_port_start(dev, 1); dma2[idx] = pnp_dma(dev, 1); } + snd_card_set_dev(card, &pcard->card->dev); ret = create_sscape(idx, card); if (ret < 0) goto _release_card; - snd_card_set_dev(card, &pcard->card->dev); if ((ret = snd_card_register(card)) < 0) { printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; -- 1.6.0.3 ---------------------------------------------------------------------- Nie dla nich ciepla praca za biurkiem. Sprawdz >>> http://link.interia.pl/f2383

2 1

[alsa-devel] [PATCH] ASoC: DaVinci: McASP FIFO related updates
by Chaithrika U S 01 Oct '09

01 Oct '09

The DMA params for McASP with FIFO has been updated so that it works for various FIFO levels. A member- 'fifo_level' has been added to the DMA params data structure. The fifo_level can be adjusted by the tx[rx]_numevt platform data. This is relevant only for DA8xx/OMAP-L1xx platforms. This implementation has been tested for numevt values 1, 2, 4, 8. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies to ALSA GIT tree on branch topic/asoc at http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=shortlog; h=topic/asoc sound/soc/davinci/davinci-i2s.c | 2 ++ sound/soc/davinci/davinci-mcasp.c | 17 +++++++---------- sound/soc/davinci/davinci-pcm.c | 21 ++++++++++++++++++--- sound/soc/davinci/davinci-pcm.h | 1 + 4 files changed, 28 insertions(+), 13 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 4ae7070..2ab8093 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -397,6 +397,8 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, } dma_params->acnt = dma_params->data_type; + dma_params->fifo_level = 0; + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(1); xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(1); diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 5d1f98a..50ad051 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -714,16 +714,13 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, struct davinci_pcm_dma_params *dma_params = &dev->dma_params[substream->stream]; int word_length; - u8 numevt; + u8 fifo_level; davinci_hw_common_param(dev, substream->stream); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - numevt = dev->txnumevt; + fifo_level = dev->txnumevt; else - numevt = dev->rxnumevt; - - if (!numevt) - numevt = 1; + fifo_level = dev->rxnumevt; if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) davinci_hw_dit_param(dev); @@ -751,12 +748,12 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - if (dev->version == MCASP_VERSION_2) { - dma_params->data_type *= numevt; - dma_params->acnt = 4 * numevt; - } else + if (dev->version == MCASP_VERSION_2 && !fifo_level) + dma_params->acnt = 4; + else dma_params->acnt = dma_params->data_type; + dma_params->fifo_level = fifo_level; davinci_config_channel_size(dev, word_length); return 0; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 359e99e..1152d8b 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -66,38 +66,53 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) dma_addr_t dma_pos; dma_addr_t src, dst; unsigned short src_bidx, dst_bidx; + unsigned short src_cidx, dst_cidx; unsigned int data_type; unsigned short acnt; unsigned int count; + unsigned int fifo_level; period_size = snd_pcm_lib_period_bytes(substream); dma_offset = prtd->period * period_size; dma_pos = runtime->dma_addr + dma_offset; + fifo_level = prtd->params->fifo_level; pr_debug("davinci_pcm: audio_set_dma_params_play channel = %d " "dma_ptr = %x period_size=%x\n", lch, dma_pos, period_size); data_type = prtd->params->data_type; count = period_size / data_type; + if (fifo_level) + count /= fifo_level; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { src = dma_pos; dst = prtd->params->dma_addr; src_bidx = data_type; dst_bidx = 0; + src_cidx = data_type * fifo_level; + dst_cidx = 0; } else { src = prtd->params->dma_addr; dst = dma_pos; src_bidx = 0; dst_bidx = data_type; + src_cidx = 0; + dst_cidx = data_type * fifo_level; } acnt = prtd->params->acnt; edma_set_src(lch, src, INCR, W8BIT); edma_set_dest(lch, dst, INCR, W8BIT); - edma_set_src_index(lch, src_bidx, 0); - edma_set_dest_index(lch, dst_bidx, 0); - edma_set_transfer_params(lch, acnt, count, 1, 0, ASYNC); + + edma_set_src_index(lch, src_bidx, src_cidx); + edma_set_dest_index(lch, dst_bidx, dst_cidx); + + if (!fifo_level) + edma_set_transfer_params(lch, acnt, count, 1, 0, ASYNC); + else + edma_set_transfer_params(lch, acnt, fifo_level, count, + fifo_level, ABSYNC); prtd->period++; if (unlikely(prtd->period >= runtime->periods)) diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index 8746606..c8b0d2b 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -23,6 +23,7 @@ struct davinci_pcm_dma_params { enum dma_event_q eventq_no; /* event queue number */ unsigned char data_type; /* xfer data type */ unsigned char convert_mono_stereo; + unsigned int fifo_level; }; -- 1.5.6

3 3

[alsa-devel] ALC268 - Acer Aspire 5720 - mic source switch disappered
by Maxim Levitsky 30 Sep '09

30 Sep '09

Hi, I have installed 2.6.32-rc1 (well -rc2...) And I notice that 'Input source' is missing. /proc/asound/Intel/codec#0 attached. Best regards, Maxim Levitsky

2 7

[PATCH] ascenario: Add scenario support to alsa-lib
by Stefan Schmidt 30 Sep '09

30 Sep '09

It allows switching audio settings between scenarios or uses-cases like listening to music and answering an incoming phone call. Made of control aliasing for playback, capture master and switch as well as the option to post- and prefix a sequence of control changes avoiding pops and other unwanted noise. Some example programs will be available in alsa-utils. CC: Ian Molton <ian(a)mnementh.co.uk> CC: Graeme Gregory <gg(a)slimlogic.co.uk> Signed-off-by: Liam Girdwood <lrg(a)slimlogic.co.uk> Signed-off-by: Stefan Schmidt <stefan(a)slimlogic.co.uk> --- include/Makefile.am | 2 +- include/ascenario.h | 170 +++++++ src/Makefile.am | 2 +- src/ascenario.c | 1360 +++++++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 1532 insertions(+), 2 deletions(-) create mode 100644 include/ascenario.h create mode 100644 src/ascenario.c diff --git a/include/Makefile.am b/include/Makefile.am index a291503..572fbc9 100644 --- a/include/Makefile.am +++ b/include/Makefile.am @@ -3,7 +3,7 @@ SUBDIRS = sound sysincludedir = ${includedir}/sys alsaincludedir = ${includedir}/alsa -alsainclude_HEADERS = asoundlib.h asoundef.h \ +alsainclude_HEADERS = asoundlib.h asoundef.h ascenario.h \ version.h global.h input.h output.h error.h \ conf.h control.h iatomic.h diff --git a/include/ascenario.h b/include/ascenario.h new file mode 100644 index 0000000..869f2ea --- /dev/null +++ b/include/ascenario.h @@ -0,0 +1,170 @@ +/* +* ALSA Scenario header file +* +* This library is free software; you can redistribute it and/or modify +* it under the terms of the GNU Lesser General Public License as +* published by the Free Software Foundation; either version 2.1 of +* the License, or (at your option) any later version. +* +* This program is distributed in the hope that it will be useful, +* but WITHOUT ANY WARRANTY; without even the implied warranty of +* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +* GNU Lesser General Public License for more details. +* +* You should have received a copy of the GNU Lesser General Public +* License along with this library; if not, write to the Free Software +* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA +* +* Copyright (C) 2008-2009 SlimLogic Ltd +* Authors: Liam Girdwood <lrg(a)slimlogic.co.uk> +* Stefan Schmidt <stefan(a)slimlogic.co.uk> +*/ + +/** + * Scenario IDs + * + * Standard Scenario IDs - Add new scenarios at the end. + */ + +#define SND_SCN_PLAYBACK_SPEAKER "playback speaker" +#define SND_SCN_PLAYBACK_HEADPHONES "playback headphone" +#define SND_SCN_PLAYBACK_HEADSET "playback headset" +#define SND_SCN_PLAYBACK_BLUETOOTH "playback bluetooth" +#define SND_SCN_PLAYBACK_HANDSET "playback handset" +#define SND_SCN_PLAYBACK_GSM "playback gsm" +#define SND_SCN_PLAYBACK_LINE "playback line" + +#define SND_SCN_CAPTURE_MIC "capture mic" +#define SND_SCN_CAPTURE_LINE "capture line" +#define SND_SCN_CAPTURE_HEADSET "capture headset" +#define SND_SCN_CAPTURE_HANDSET "capture handset" +#define SND_SCN_CAPTURE_BLUETOOTH "capture bluetooth" +#define SND_SCN_CAPTURE_GSM "capture gsm" + +#define SND_SCN_PHONECALL_GSM_HANDSET "phonecall gsm handset" +#define SND_SCN_PHONECALL_BT_HANDSET "phonecall bt handset" +#define SND_SCN_PHONECALL_IP_HANDSET "phonecall ip handset" +#define SND_SCN_PHONECALL_GSM_HEADSET "phonecall gsm headset" +#define SND_SCN_PHONECALL_BT_HEADSET "phonecall bt headset" +#define SND_SCN_PHONECALL_IP_HEADSET "phonecall ip headset" + +/** + * QOS + * + * Defines Audio Quality of Service. Systems supporting different types of QoS + * often have lower power consumption on lower quality levels. + */ +#define SND_POWER_QOS_HIFI 0 +#define SND_POWER_QOS_VOICE 1 +#define SND_POWER_QOS_SYSTEM 2 + +struct snd_scenario; + +/* TODO: add notification */ + +/** + * snd_scenario_list - list supported scenarios + * @scn: scenario + * @list: list of supported scenario names. + * + * List supported scenarios for this sound card. + * Returns number of scenarios. + */ +int snd_scenario_list(struct snd_scenario *scn, const char **list[]); + +/** + * snd_scenario_set_scn - set scenario + * @scn: scenario + * @scenario: scenario name + * + * Set new scenario for sound card. + */ +int snd_scenario_set_scn(struct snd_scenario *scn, const char *scenario); + +/** + * snd_scenario_get_scn - get scenario + * @scn: scenario + * + * Get current sound card scenario. + */ +const char *snd_scenario_get_scn(struct snd_scenario *scn); + +/** + * snd_scenario_get_master_playback_volume - get playback volume + * @scn: scenario + * + * Get the master playback volume control name for the current scenario. + */ +int snd_scenario_get_master_playback_volume(struct snd_scenario *scn); + +/** + * snd_scenario_get_master_playback_switch - get playback switch + * @scn: scenario + * + * Get the master playback switch control name for the current scenario. + */ + int snd_scenario_get_master_playback_switch(struct snd_scenario *scn); + +/** + * snd_scenario_get_master_capture_volume - get capture volume + * @scn: scenario + * + * Get the master capture volume control name for the current scenario. + */ +int snd_scenario_get_master_capture_volume(struct snd_scenario *scn); + +/** + * snd_scenario_get_master_capture_switch - get capture switch + * @scn: scenario + * + * Get the master capture switch control name for the current scenario. + */ +int snd_scenario_get_master_capture_switch(struct snd_scenario *scn); + +/** + * snd_scenario_set_qos - set qos + * @qos: qos + * + * Set Quality of Service for this scenario. + */ +int snd_scenario_set_qos(struct snd_scenario *scn, int qos); + +/** + * snd_scenario_get_qos - get qos + * @scn: scenario + * + * Get Quality of Service for this scenario. + */ +int snd_scenario_get_qos(struct snd_scenario *scn); + +/** + * snd_scenario_open - open scenario core + * @card_name: sound card name. + * + * Open scenario manager core for sound card. + */ +struct snd_scenario *snd_scenario_open(const char *card_name); + +/** + * snd_scenario_reload - reload and reparse scenario configuration + * @scn: scenario + * + * Reloads and reparses sound card scenario configuration. + */ +int snd_scenario_reload(struct snd_scenario *scn); + +/** + * snd_scenario_close - close scenario core + * @scn: scenario + * + * Free scenario manager core for sound card. + */ +void snd_scenario_close(struct snd_scenario *scn); + +/** + * snd_scenario_dump - dump + * @card_name: sound card name. + * + * Dump current sound card settings to stdout in scn format. + */ +int snd_scenario_dump(const char *card_name); diff --git a/src/Makefile.am b/src/Makefile.am index 3204fe4..be46cb3 100644 --- a/src/Makefile.am +++ b/src/Makefile.am @@ -14,7 +14,7 @@ SYMFUNCS = endif lib_LTLIBRARIES = libasound.la -libasound_la_SOURCES = conf.c confmisc.c input.c output.c async.c error.c dlmisc.c socket.c shmarea.c userfile.c names.c +libasound_la_SOURCES = conf.c confmisc.c input.c output.c async.c error.c dlmisc.c socket.c shmarea.c userfile.c names.c ascenario.c SUBDIRS=control libasound_la_LIBADD = control/libcontrol.la diff --git a/src/ascenario.c b/src/ascenario.c new file mode 100644 index 0000000..99f239e --- /dev/null +++ b/src/ascenario.c @@ -0,0 +1,1360 @@ +/* + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. + * + * Copyright (C) 2008-2009 SlimLogic Ltd + * Authors: Liam Girdwood <lrg(a)slimlogic.co.uk> + * Stefan Schmidt <stefan(a)slimlogic.co.uk> + */ + +#define _GNU_SOURCE /* needed of O_NOATIME */ + +#include <stdio.h> +#include <unistd.h> +#include <errno.h> +#include <string.h> +#include <sys/types.h> +#include <sys/stat.h> +#include <fcntl.h> +#include <ctype.h> +#include <alsa/asoundlib.h> + +#include "../include/ascenario.h" + +#define PRE_SEQ 0 +#define POST_SEQ 1 +#define MAX_SCN 32 +#define MAX_NAME 64 +#define MAX_FILE 256 +#define MAX_BUF 256 +#define ALSA_SCN_DIR "/etc/alsa/scenario" + +/* + * Stores all scenario settings for 1 kcontrol. Hence we have a + * control_settings for each kcontrol in card. + */ +struct control_settings { + char name[MAX_NAME]; + int id; + snd_ctl_elem_type_t type; + int count; /* 1 = mono, 2 = stereo, etc */ + unsigned short *value; /* kcontrol value 2D array */ +}; + +/* + * If sleep is 0 the element contains the settings in control. Else sleep + * contains the sleep time in micro seconds. + */ +struct sequence_element { + unsigned int sleep; /* Sleep time in msecs if sleep element, else 0 */ + struct control_settings *control; + struct sequence_element *next; /* Pointer to next list element */ +}; + +/* + * Describes default mixers and qos for scenario. + * We have a scenario_info for each scenario loaded. + */ +struct scenario_info { + char *name; + char *file; + char *pre_sequence_file; + char *post_sequence_file; + short playback_volume_id; + short playback_switch_id; + short capture_volume_id; + short capture_switch_id; + int qos; +}; + +/* Describe a snd card and all its scenarios. + */ +struct snd_scenario { + char *card_name; + int current_scenario; + int num_scenarios; /* number of supported scenarios */ + int num_kcontrols; /* number of kcontrols */ + struct sequence_element *pre_seq_list; /* Linked list for pre sequence */ + struct sequence_element *post_seq_list; /* Linked list for post sequence */ + const char **list; + struct scenario_info *scenario; /* var len array of scenario info */ + struct control_settings *control; /* var len array of controls */ +}; + +static void scn_error(const char *fmt,...) +{ + va_list va; + va_start(va, fmt); + fprintf(stderr, "scenario: "); + vfprintf(stderr, fmt, va); + va_end(va); +} + +static void scn_stdout(const char *fmt,...) +{ + va_list va; + va_start(va, fmt); + vfprintf(stdout, fmt, va); + va_end(va); +} + +static inline void set_value(struct snd_scenario *scn, + struct control_settings *control, int count, unsigned short val) +{ + int offset = scn->current_scenario * control->count; + control->value[offset + count] = val; +} + +static inline unsigned short get_value(struct snd_scenario *scn, + struct control_settings *control, int count) +{ + int offset = scn->current_scenario * control->count; + return control->value[offset + count]; +} + +static int dump_control(snd_ctl_t *handle, snd_ctl_elem_id_t *id) +{ + int err, count, i; + snd_ctl_elem_info_t *info; + snd_ctl_elem_type_t type; + snd_ctl_elem_value_t *control; + + snd_ctl_elem_info_alloca(&info); + snd_ctl_elem_value_alloca(&control); + + snd_ctl_elem_info_set_id(info, id); + err = snd_ctl_elem_info(handle, info); + if (err < 0) { + scn_stdout("%s: failed to get ctl info\n"); + return err; + } + + snd_ctl_elem_value_set_id(control, id); + snd_ctl_elem_read(handle, control); + + type = snd_ctl_elem_info_get_type(info); + count = snd_ctl_elem_info_get_count(info); + if (count == 0) + return 0; + + scn_stdout("%u:'%s':%d:", + snd_ctl_elem_id_get_numid(id), + snd_ctl_elem_id_get_name(id), count); + + switch (type) { + case SND_CTL_ELEM_TYPE_BOOLEAN: + for (i = 0; i < count - 1; i++) + scn_stdout("%d,", + snd_ctl_elem_value_get_boolean(control, i)); + scn_stdout("%d", snd_ctl_elem_value_get_boolean(control, i)); + break; + case SND_CTL_ELEM_TYPE_INTEGER: + for (i = 0; i < count - 1; i++) + scn_stdout("%d,", + snd_ctl_elem_value_get_integer(control, i)); + scn_stdout("%d", snd_ctl_elem_value_get_integer(control, i)); + break; + case SND_CTL_ELEM_TYPE_INTEGER64: + for (i = 0; i < count - 1; i++) + scn_stdout("%ld,", + snd_ctl_elem_value_get_integer64(control, i)); + scn_stdout("%ld", + snd_ctl_elem_value_get_integer64(control, i)); + break; + case SND_CTL_ELEM_TYPE_ENUMERATED: + for (i = 0; i < count - 1; i++) + scn_stdout("%d,", + snd_ctl_elem_value_get_enumerated(control, i)); + scn_stdout("%d", + snd_ctl_elem_value_get_enumerated(control, i)); + break; + case SND_CTL_ELEM_TYPE_BYTES: + for (i = 0; i < count - 1; i++) + scn_stdout("%2.2x,", + snd_ctl_elem_value_get_byte(control, i)); + scn_stdout("%2.2x", snd_ctl_elem_value_get_byte(control, i)); + break; + default: + break; + } + scn_stdout("\n"); + return 0; +} + +/* + * Add new kcontrol from sound card into memory database. + */ +static int add_control(snd_ctl_t *handle, snd_ctl_elem_id_t *id, + struct control_settings *control_settings) +{ + int err; + snd_ctl_elem_info_t *info; + snd_ctl_elem_value_t *control; + + snd_ctl_elem_info_alloca(&info); + snd_ctl_elem_value_alloca(&control); + + snd_ctl_elem_info_set_id(info, id); + err = snd_ctl_elem_info(handle, info); + if (err < 0) { + scn_stdout("%s: failed to get ctl info\n"); + return err; + } + + snd_ctl_elem_value_set_id(control, id); + snd_ctl_elem_read(handle, control); + + strncpy(control_settings->name, snd_ctl_elem_id_get_name(id), + MAX_NAME); + control_settings->count = snd_ctl_elem_info_get_count(info); + control_settings->type = snd_ctl_elem_info_get_type(info); + control_settings->id = snd_ctl_elem_id_get_numid(id); + return 0; +} + +static int parse_controls(struct snd_scenario *scn, FILE *f) +{ + struct control_settings *control; + char buf[MAX_BUF], name[MAX_NAME]; + int id, count, line = 1, i; + char *name_start, *name_end, *tbuf; + + while (fgets(buf, MAX_BUF, f) != NULL) { + + /* get id */ + tbuf = buf; + id = atoi(tbuf); + if (id == 0) { + scn_error("%s:id == 0 on line %d\n", __func__, line); + return -EINVAL; + } + for (i = 0; i < scn->num_kcontrols; i++) { + if (id == scn->control[i].id) { + control = &scn->control[i]; + goto get_name; + } + } + scn_error("%s:id not found at line %d\n", __func__, line); + return -EINVAL; +get_name: + /* get name start */ + while (*tbuf != 0 && *tbuf != '\'') + tbuf++; + if (*tbuf == 0) + return -EINVAL; + name_start = ++tbuf; + + /* get name end */ + while (*tbuf != 0 && *tbuf != '\'') + tbuf++; + if (*tbuf == 0) + return -EINVAL; + name_end = tbuf++; + + /* copy name */ + if ((name_end - name_start) > MAX_NAME) { + scn_error("%s:name too big at %d chars line %d\n", + __func__, name_end - name_start, line); + return -EINVAL; + } + strncpy(name, name_start, name_end - name_start); + name[name_end - name_start] = 0; + if (strcmp(name, control->name)) { + scn_error("%s: name %s and %s don't match at line %d\n", + __func__, name, control->name, line); + return -EINVAL; + } + + /* get count */ + tbuf++; + count = atoi(tbuf); + if (count == 0) { + scn_error("%s:count == 0 on line %d\n", __func__, + line); + return -EINVAL; + } + if (count != control->count) { + scn_error("%s:count does not match at line %d\n", + __func__, line); + return -EINVAL; + } + + /* get vals */ + control->value = malloc(control->count * scn->num_scenarios * + sizeof(unsigned short)); + if (control->value == NULL) + return -ENOMEM; + + while (*tbuf != 0 && *tbuf != ':') + tbuf++; + if (*tbuf == 0) + return -EINVAL; + tbuf++; + + for (i = 0; i < count; i++) { + set_value(scn, control, i, atoi(tbuf)); + while (*tbuf != 0 && *tbuf != ',') + tbuf++; + + if (*tbuf++ == 0 && i < (count - 1)) + return -EINVAL; + } + line++; + } + + return 0; +} + +static char *get_string (char *buf) +{ + char *str, *end; + + /* find '=' */ + while (isblank(*buf)) + buf++; + if (*buf == 0 || *buf != '=') { + scn_error("%s: missing '='\n", __func__); + return NULL; + } + + /* find leading '"' */ + buf++; + while (isblank(*buf)) + buf++; + if (*buf == 0 || *buf != '"') { + scn_error("%s: missing start '\"'\n", __func__); + return NULL; + } + str = ++buf; + + /* get value */ + while (*buf != 0 && *buf != '"') + buf++; + end = buf; + + /* find '"' terminator */ + if (*buf == 0 || *buf != '"') { + scn_error("%s: missing terminator '\"' %s\n", __func__, buf); + return NULL; + } + + *end = 0; + return strdup(str); +} + +static char *get_control_name (char *buf) +{ + char *str, *end; + + /* find leading '"' */ + buf++; + while (isblank(*buf)) + buf++; + if (*buf == 0 || *buf != '"') { + scn_error("%s: missing start '\"'\n", __func__); + return NULL; + } + str = ++buf; + + /* get value */ + while (*buf != 0 && *buf != '"') + buf++; + end = buf; + + /* find '"' terminator */ + if (*buf == 0 || *buf != '"') { + scn_error("%s: missing terminator '\"' %s\n", __func__, buf); + return NULL; + } + + *end = 0; + return strdup(str); +} + +static int get_int (char *buf) +{ + /* find '=' */ + while (isblank(*buf)) + buf++; + if (*buf == 0 || *buf != '=') { + scn_error("%s: missing '='\n", __func__); + return -EINVAL; + } + buf++; + return atoi(buf); +} + +static int get_enum (char *buf) +{ + /* find '=' */ + while (isblank(*buf)) + buf++; + if (*buf == 0 || *buf != '=') { + scn_error("%s: missing '='\n", __func__); + return -EINVAL; + } + buf++; + return 0; /* TODO */ +} + +static void seq_list_append(struct snd_scenario *scn, + struct sequence_element *curr, int position) +{ + struct sequence_element *last, *tmp; + + if (position) { + if (!scn->post_seq_list) + scn->post_seq_list = curr; + + else { + tmp = scn->post_seq_list; + while (tmp) { + last = tmp; + tmp = tmp->next; + } + last->next = curr; + } + } + else { + if (!scn->pre_seq_list) { + scn->pre_seq_list = curr; + } + else { + tmp = scn->pre_seq_list; + while (tmp) { + last = tmp; + tmp = tmp->next; + } + last->next = curr; + } + } +} + +static int parse_sequences(struct snd_scenario *scn, FILE *f, int position) +{ + char buf[MAX_BUF], *tbuf, *control_value; + int control_len, i; + struct sequence_element *curr; + + while (fgets(buf, MAX_BUF, f) != NULL) { + + /* Check for lines with comments and ignore */ + if (buf[0] == '#') + continue; + + /* Parse current line and skip blanks */ + tbuf = buf; + while (isblank(*tbuf)) + tbuf++; + + curr = malloc(sizeof(struct sequence_element)); + if (curr == NULL) + return -ENOMEM; + bzero(curr, sizeof(struct sequence_element)); + + curr->control = malloc(sizeof(struct control_settings)); + if (curr->control == NULL) + return -ENOMEM; + bzero(curr->control, sizeof(struct control_settings)); + + curr->control->value = malloc(curr->control->count * scn->num_scenarios + * sizeof(unsigned short)); + if (curr->control->value == NULL) + return -ENOMEM; + bzero(curr->control->value, curr->control->count * scn->num_scenarios + * sizeof(unsigned short)); + + if (strncmp(tbuf, "kcontrol", 8) == 0) { + strncpy(curr->control->name, get_control_name(tbuf + 8), MAX_NAME); + control_len = strlen(curr->control->name); + /* 11 = 8 from kcontrol + 2 quotes + 1 blank */ + control_value = get_string(tbuf + 11 + control_len); + + for (i = 0; i < scn->num_kcontrols; i++) { + if (strncmp(curr->control->name, scn->control[i].name, + control_len) == 0) { + curr->sleep = 0; + curr->control->id = scn->control[i].id; + curr->control->type = scn->control[i].type; + curr->control->count = scn->control[i].count; + set_value(scn, curr->control, curr->control->count, + atoi(control_value)); + seq_list_append(scn, curr, position); + } + } + + continue; + } + + if (strncmp(tbuf, "msleep", 6) == 0) { + curr->sleep = get_int(tbuf + 6); + + /* Free control elements as we only have a sleep element + * here */ + if (curr->control) { + if (curr->control->value) + free(curr->control->value); + free(curr->control); + } + + seq_list_append(scn, curr, position); + continue; + } + } + + return 0; +} + +/* load scenario i */ +static int read_scenario_file(struct snd_scenario *scn) +{ + int fd, ret; + FILE *f; + char filename[MAX_FILE]; + struct scenario_info *info = &scn->scenario[scn->current_scenario]; + + snprintf(filename, MAX_FILE, "%s/%s/%s", ALSA_SCN_DIR, scn->card_name, + info->file); + + fd = open(filename, O_RDONLY | O_NOATIME); + if (fd < 0) { + scn_error("%s: couldn't open %s\n", __func__, filename); + return fd; + } + + f = fdopen(fd, "r"); + if (f == NULL) { + ret = errno; + goto close; + } + + ret = parse_controls(scn, f); + fclose(f); +close: + close(fd); + return ret; +} + +static int read_sequence_file(struct snd_scenario *scn, int position) +{ + int fd, ret; + FILE *f; + char filename[MAX_FILE]; + struct scenario_info *info = &scn->scenario[scn->current_scenario]; + + if (position == PRE_SEQ) { + sprintf(filename, "%s/%s/%s", ALSA_SCN_DIR, scn->card_name, + info->pre_sequence_file); + } + else { + sprintf(filename, "%s/%s/%s", ALSA_SCN_DIR, scn->card_name, + info->post_sequence_file); + } + + fd = open(filename, O_RDONLY | O_NOATIME); + if (fd < 0) { + return fd; + } + + f = fdopen(fd, "r"); + if (f == NULL) { + ret = errno; + goto close; + } + + ret = parse_sequences(scn, f, position); + fclose(f); +close: + close(fd); + return ret; +} + +static int parse_scenario(struct snd_scenario *scn, FILE *f, int line_) +{ + struct scenario_info *info; + int line = line_ - 1, id = 0, file = 0, len, offset; + char buf[MAX_BUF], *tbuf; + + offset = sizeof(scn->scenario); + scn->scenario = realloc(scn->scenario, + (scn->num_scenarios + 1) * sizeof(struct scenario_info)); + len = sizeof(scn->scenario) - offset; + if (scn->scenario == NULL) + return -ENOMEM; + bzero(scn->scenario + offset, len); + info = scn->scenario + scn->num_scenarios; + + /* Set sequence filename to NULL as it is optional and we want to check + * for NULL to avoid segfaults */ + info->pre_sequence_file = NULL; + info->post_sequence_file = NULL; + + while(fgets(buf, MAX_BUF, f) != NULL) { + + line++; + if (buf[0] == '#') + continue; + + tbuf = buf; + while (isblank(*tbuf)) + tbuf++; + + if (strncmp(tbuf, "Identifier", 10) == 0) { + info->name = get_string(tbuf + 10); + if (info->name == NULL) { + scn_error("%s: failed to get Identifer\n", + __func__); + goto err; + } + id = 1; + continue; + } + + if (strncmp(tbuf, "File", 4) == 0) { + info->file = get_string(tbuf + 4); + if (info->file == NULL) { + scn_error("%s: failed to get File\n", + __func__); + goto err; + } + file = 1; + continue; + } + + if (strncmp(tbuf, "QoS", 3) == 0) { + info->qos = get_enum(tbuf + 3); + if (info->qos < 0) { + scn_error("%s: failed to get QoS\n", + __func__); + goto err; + } + continue; + } + + if (strncmp(tbuf, "Master Playback Volume", 22) == 0) { + info->playback_volume_id = get_int(tbuf + 22); + if (info->playback_volume_id < 0) { + scn_error("%s: failed to get Master Playback Volume\n", + __func__); + goto err; + } + continue; + } + + if (strncmp(tbuf, "Master Playback Switch", 22) == 0) { + info->playback_switch_id = get_int(tbuf + 22); + if (info->playback_switch_id < 0) { + scn_error("%s: failed to get Master Playback Switch\n", + __func__); + goto err; + } + continue; + } + + if (strncmp(tbuf, "Master Capture Volume", 21) == 0) { + info->capture_volume_id = get_int(tbuf + 21); + if (info->capture_volume_id < 0) { + scn_error("%s: failed to get Master Capture Volume\n", + __func__); + goto err; + } + continue; + } + + if (strncmp(tbuf, "Master Capture Switch", 21) == 0) { + info->capture_switch_id = get_int(tbuf + 21); + if (info->capture_switch_id < 0) { + scn_error("%s: failed to get Master Capture Switch\n", + __func__); + goto err; + } + continue; + } + + if (strncmp(tbuf, "PreSequenceFile", 15) == 0) { + info->pre_sequence_file = get_string(tbuf + 15); + if (info->pre_sequence_file == NULL) { + scn_error("%s: failed to get PreSequenceFile\n", + __func__); + goto err; + } + continue; + } + + if (strncmp(tbuf, "PostSequenceFile", 16) == 0) { + info->post_sequence_file = get_string(tbuf + 16); + if (info->post_sequence_file == NULL) { + scn_error("%s: failed to get PostSequenceFile\n", + __func__); + goto err; + } + continue; + } + + if (strncmp(tbuf, "EndSection", 10) == 0) { + break; + } + } + + if (file & id) { + scn->num_scenarios++; + return 0; + } +err: + if (file) { + free(info->file); + info->file = NULL; + } + if (id) { + free(info->name); + info->name = NULL; + } + return -EINVAL; +} + +static int read_master_file(struct snd_scenario *scn, FILE *f) +{ + int line = 0, ret = 0, i; + char buf[MAX_BUF], *tbuf; + + /* parse master config sections */ + while(fgets(buf, MAX_BUF, f) != NULL) { + + if (buf[0] == '#') { + line++; + continue; + } + + if (strncmp(buf, "Section", 7) == 0) { + + tbuf = buf + 7; + while (isblank(*tbuf)) + tbuf++; + + if (strncmp(tbuf, "\"Scenario\"", 10) == 0) { + line = parse_scenario(scn, f, line); + if (line < 0) { + scn_error("%s: failed to parse " + "scenario\n", __func__); + goto err; + } + continue; + } + } + line++; + } + + /* copy ptrs to scenario names */ + scn->list = malloc(scn->num_scenarios * sizeof(char *)); + if (scn->list == NULL) + ret = -ENOMEM; + for (i = 0; i < scn->num_scenarios; i++) + scn->list[i] = scn->scenario[i].name; + +err: + if (ferror(f)) { + scn_error("%s: failed to read master\n", __func__); + return ferror(f); + } + return ret; +} + +/* load scenario i */ +static int import_master_config(struct snd_scenario *scn) +{ + int fd, ret; + FILE *f; + char filename[MAX_FILE]; + + sprintf(filename, "%s/%s.conf", ALSA_SCN_DIR, scn->card_name); + + fd = open(filename, O_RDONLY | O_NOATIME); + if (fd < 0) { + scn_error("%s: couldn't open %s\n", __func__, filename); + return fd; + } + + f = fdopen(fd, "r"); + if (f == NULL) { + ret = errno; + goto close; + } + + ret = read_master_file(scn, f); + fclose(f); +close: + close(fd); + return ret; +} + +/* parse_card_controls + * @scn: scenario + * + * Parse sound card and store control data in memory db. + */ +static int parse_card_controls(struct snd_scenario *scn) +{ + struct control_settings *control; + snd_ctl_t *handle; + snd_ctl_card_info_t *info; + snd_ctl_elem_list_t *list; + int ret, i; + + snd_ctl_card_info_alloca(&info); + snd_ctl_elem_list_alloca(&list); + + /* open and load snd card */ + ret = snd_ctl_open(&handle, scn->card_name, SND_CTL_READONLY); + if (ret < 0) { + scn_error("%s: control %s open retor: %s\n", __func__, + scn->card_name, snd_strerror(ret)); + return ret; + } + + ret = snd_ctl_card_info(handle, info); + if (ret < 0) { + scn_error("%s :control %s local retor: %s\n", __func__, + scn->card_name, snd_strerror(ret)); + goto close; + } + + ret = snd_ctl_elem_list(handle, list); + if (ret < 0) { + scn_error("%s: cannot determine controls: %s\n", __func__, + snd_strerror(ret)); + goto close; + } + + scn->num_kcontrols = snd_ctl_elem_list_get_count(list); + if (scn->num_kcontrols < 0) { + ret = 0; + goto close; + } + + snd_ctl_elem_list_set_offset(list, 0); + if (snd_ctl_elem_list_alloc_space(list, scn->num_kcontrols) < 0) { + scn_error("%s: not enough memory...\n", __func__); + ret = -ENOMEM; + goto close; + } + if ((ret = snd_ctl_elem_list(handle, list)) < 0) { + scn_error("%s: cannot determine controls: %s\n", __func__, + snd_strerror(ret)); + goto free; + } + + /* allocate db memory for controls */ + scn->control = calloc(scn->num_kcontrols, + sizeof(struct control_settings)); + if (scn->control == NULL) { + ret = -ENOMEM; + goto free; + } + control = scn->control; + + /* iterate through each kcontrol and add to db */ + for (i = 0; i < scn->num_kcontrols; ++i) { + snd_ctl_elem_id_t *id; + snd_ctl_elem_id_alloca(&id); + snd_ctl_elem_list_get_id(list, i, id); + + ret = add_control(handle, id, control++); + if (ret < 0) { + scn_error("%s: failed to add control error %s\n", + __func__, snd_strerror(ret)); + goto close; + } + } +free: + snd_ctl_elem_list_free_space(list); +close: + snd_ctl_close(handle); + return ret; +} + +/* import_scenario_files - + * @scn: scenario + * + * Read and parse scenario_info files the store in memory. + */ +static int import_scenario_files(struct snd_scenario *scn) +{ + int ret; + + ret = import_master_config(scn); + if (ret < 0) { + scn_error("%s: failed to parse master scenario config\n", + __func__); + return ret; + } + + for (scn->current_scenario = 0; + scn->current_scenario < scn->num_scenarios; + scn->current_scenario++) { + + ret = read_scenario_file(scn); + if (ret < 0) { + scn_error("%s: failed to parse scenario %s\n", + __func__, + scn->scenario[scn->current_scenario].name); + scn->current_scenario = -1; + return ret; + } + + if (scn->scenario[scn->current_scenario].pre_sequence_file != NULL) { + ret = read_sequence_file(scn, PRE_SEQ); + if (ret < 0) { + scn_stdout("Warning: PreSequence file defined but" + " missing in scenario \"%s\"\n", + scn->scenario[scn->current_scenario].name); + } + } + + if (scn->scenario[scn->current_scenario].post_sequence_file != NULL) { + ret = read_sequence_file(scn, POST_SEQ); + if (ret < 0) { + scn_stdout("Warning: PostSequence file defined but" + " missing in scenario \"%s\"\n", + scn->scenario[scn->current_scenario].name); + } + } + + } + return 0; +} + +/* free all resorces */ +static void free_scn(struct snd_scenario *scn) +{ + /* TODO: valgrind to make sure. */ + int i; + + if (scn == NULL) + return; + + if (scn->control) { + if (scn->control->value) + free(scn->control->value); + free(scn->control); + } + + if (scn->list) + free(scn->list); + if (scn->card_name) + free(scn->card_name); + if (scn->pre_seq_list) + free(scn->pre_seq_list); + if (scn->post_seq_list) + free(scn->post_seq_list); + + if (scn->scenario) { + for (i = 0; i < scn->num_scenarios; i++) { + struct scenario_info *info = &scn->scenario[i]; + + if (info->name) + free(info->name); + if (info->file) + free(info->file); + if (info->pre_sequence_file) + free(info->pre_sequence_file); + if (info->post_sequence_file) + free(info->post_sequence_file); + } + free(scn->scenario); + } + free(scn); +} + +/* + * Init sound card scenario db. + */ +struct snd_scenario *snd_scenario_open(const char *card_name) +{ + struct snd_scenario *scn; + int err; + + /* TODO: locking and + * check if card_name scn is already loaded, + * if so reuse to conserve ram. */ + + scn = malloc(sizeof(struct snd_scenario)); + if (scn == NULL) + return NULL; + bzero(scn, sizeof(struct snd_scenario)); + scn->card_name = strdup(card_name); + if (scn->card_name == NULL) { + free(scn); + return NULL; + } + + /* get info about sound card */ + err = parse_card_controls(scn); + if (err < 0) { + free_scn(scn); + return NULL; + } + + /* get info on scenarios and verify against card */ + err = import_scenario_files(scn); + if (err < 0) { + free_scn(scn); + return NULL; + } + + return scn; +} + +/* + * Reload and reparse scenario db. + */ +int snd_scenario_reload(struct snd_scenario *scn) +{ + free_scn(scn); + + scn->num_kcontrols = parse_card_controls(scn); + if (scn->num_kcontrols <= 0) { + free_scn(scn); + return -EINVAL; + } + + scn->num_scenarios = import_scenario_files(scn); + if (scn->num_scenarios <= 0) { + return -EINVAL; + } + + return 0; +} + +void snd_scenario_close(struct snd_scenario *scn) +{ + free_scn(scn); +} + +static int set_control(snd_ctl_t *handle, snd_ctl_elem_id_t *id, + struct snd_scenario *scn) +{ + struct control_settings *setting; + int ret, count, i, idnum; + snd_ctl_elem_info_t *info; + snd_ctl_elem_type_t type; + snd_ctl_elem_value_t *control; + + snd_ctl_elem_info_alloca(&info); + snd_ctl_elem_value_alloca(&control); + + snd_ctl_elem_info_set_id(info, id); + ret = snd_ctl_elem_info(handle, info); + if (ret < 0) { + scn_error("%s: failed to get ctl info\n", __func__); + return ret; + } + + snd_ctl_elem_value_set_id(control, id); + snd_ctl_elem_read(handle, control); + + idnum = snd_ctl_elem_id_get_numid(id); + for (i = 0; i < scn->num_kcontrols; i++) { + setting = &scn->control[i]; + if (setting->id == idnum) + goto set_val; + } + scn_error("%s: failed to find control %d\n", __func__, idnum); + return 0; + +set_val: + type = snd_ctl_elem_info_get_type(info); + count = snd_ctl_elem_info_get_count(info); + if (count == 0) + return 0; + + switch (type) { + case SND_CTL_ELEM_TYPE_BOOLEAN: + for (i = 0; i < count; i++) + snd_ctl_elem_value_set_boolean(control, i, + get_value(scn, setting, i)); + break; + case SND_CTL_ELEM_TYPE_INTEGER: + for (i = 0; i < count; i++) + snd_ctl_elem_value_set_integer(control, i, + get_value(scn, setting, i)); + + break; + case SND_CTL_ELEM_TYPE_INTEGER64: + for (i = 0; i < count; i++) + snd_ctl_elem_value_set_integer64(control, i, + get_value(scn, setting, i)); + + break; + case SND_CTL_ELEM_TYPE_ENUMERATED: + for (i = 0; i < count; i++) + snd_ctl_elem_value_set_enumerated(control, i, + get_value(scn, setting, i)); + + break; + case SND_CTL_ELEM_TYPE_BYTES: + for (i = 0; i < count; i++) + snd_ctl_elem_value_set_byte(control, i, + get_value(scn, setting, i)); + break; + default: + break; + } + + ret = snd_ctl_elem_write(handle, control); + if (ret < 0) { + scn_error("%s: control %s failed: %s\n", __func__, + setting->name, snd_strerror(ret)); + scn_error("%s: count %d type: %d\n", __func__, + count, type); + for (i = 0; i < count; i++) + fprintf(stderr, "%d ", get_value(scn, setting, i)); + return ret; + } + return 0; +} + +static void exec_sequence(struct sequence_element *seq, struct snd_scenario + *scn, snd_ctl_elem_list_t *list, snd_ctl_t *handle) +{ + int count = snd_ctl_elem_list_get_count(list); + while (seq) { + if (seq->sleep) + usleep(seq->sleep); + else { + snd_ctl_elem_id_t *id; + snd_ctl_elem_id_alloca(&id); + int ret, i, numid; + /* Where is id lookup from numid if you need it? */ + for (i = 0; i < count; ++i) { + snd_ctl_elem_list_get_id(list, i, id); + numid = snd_ctl_elem_id_get_numid(id); + if (numid == seq->control->id) { + ret = set_control(handle, id, scn); + if (ret < 0) { + scn_error("%s: failed to set control %s\n", + __func__, scn->card_name); + } + break; + } + } + } + seq = seq->next; + } +} + +int snd_scenario_set_scn(struct snd_scenario *scn, const char *name) +{ + snd_ctl_card_info_t *info; + snd_ctl_elem_list_t *list; + snd_ctl_t *handle; + int ret, count, i; + + snd_ctl_card_info_alloca(&info); + snd_ctl_elem_list_alloca(&list); + + /* find scenario name */ + for (i = 0; i < scn->num_scenarios; i++) { + if (!strcmp(scn->scenario[i].name, name)) + goto found; + } + scn_error("%s: scenario %s not found\n", __func__, name); + return -EINVAL; + +found: + /* scenario found - now open card */ + scn->current_scenario = i; + ret = snd_ctl_open(&handle, scn->card_name, 0); + if (ret) { + scn_error("%s: control %s open error: %s\n", __func__, + scn->card_name, snd_strerror(ret)); + return ret; + } + + ret = snd_ctl_card_info(handle, info); + if (ret < 0) { + scn_error("%s :control %s local retor: %s\n", __func__, + scn->card_name, snd_strerror(ret)); + goto close; + } + + ret = snd_ctl_elem_list(handle, list); + if (ret < 0) { + scn_error("%s: cannot determine controls: %s\n", __func__, + snd_strerror(ret)); + goto close; + } + + count = snd_ctl_elem_list_get_count(list); + if (count < 0) { + ret = 0; + goto close; + } + + snd_ctl_elem_list_set_offset(list, 0); + if (snd_ctl_elem_list_alloc_space(list, count) < 0) { + scn_error("%s: not enough memory...\n", __func__); + ret = -ENOMEM; + goto close; + } + if ((ret = snd_ctl_elem_list(handle, list)) < 0) { + scn_error("%s: cannot determine controls: %s\n", __func__, + snd_strerror(ret)); + goto free; + } + + /* If we have a sequence list that should be executed before the new + * scenario is set do it now */ + if (scn->pre_seq_list) + exec_sequence(scn->pre_seq_list, scn, list, handle); + + /* iterate through each kcontrol and add to db */ + for (i = 0; i < count; ++i) { + snd_ctl_elem_id_t *id; + snd_ctl_elem_id_alloca(&id); + snd_ctl_elem_list_get_id(list, i, id); + + ret = set_control(handle, id, scn); + if (ret < 0) { + scn_error("%s: failed to set control %s\n", __func__, + scn->card_name); + } + } + + /* If we have a sequence list that should be executed after the new + * scenario is set do it now */ + if (scn->post_seq_list) + exec_sequence(scn->post_seq_list, scn, list, handle); + +free: + snd_ctl_elem_list_free_space(list); +close: + snd_ctl_close(handle); + return ret; +} + +int snd_scenario_dump(const char *card_name) +{ + snd_ctl_t *handle; + snd_ctl_card_info_t *info; + snd_ctl_elem_list_t *list; + int ret, i, count; + + snd_ctl_card_info_alloca(&info); + snd_ctl_elem_list_alloca(&list); + + /* open and load snd card */ + ret = snd_ctl_open(&handle, card_name, SND_CTL_READONLY); + if (ret < 0) { + scn_error("%s: control %s open retor: %s\n", __func__, card_name, + snd_strerror(ret)); + return ret; + } + + ret = snd_ctl_card_info(handle, info); + if (ret < 0) { + scn_error("%s :control %s local retor: %s\n", __func__, + card_name, snd_strerror(ret)); + goto close; + } + + ret = snd_ctl_elem_list(handle, list); + if (ret < 0) { + scn_error("%s: cannot determine controls: %s\n", __func__, + snd_strerror(ret)); + goto close; + } + + count = snd_ctl_elem_list_get_count(list); + if (count < 0) { + ret = 0; + goto close; + } + + snd_ctl_elem_list_set_offset(list, 0); + if (snd_ctl_elem_list_alloc_space(list, count) < 0) { + scn_error("%s: not enough memory...\n", __func__); + ret = -ENOMEM; + goto close; + } + if ((ret = snd_ctl_elem_list(handle, list)) < 0) { + scn_error("%s: cannot determine controls: %s\n", __func__, + snd_strerror(ret)); + goto free; + } + + /* iterate through each kcontrol and add to db */ + for (i = 0; i < count; ++i) { + snd_ctl_elem_id_t *id; + snd_ctl_elem_id_alloca(&id); + snd_ctl_elem_list_get_id(list, i, id); + + ret = dump_control(handle, id); + if (ret < 0) { + scn_error("%s: cannot determine controls: %s\n", + __func__, snd_strerror(ret)); + goto free; + } + } +free: + snd_ctl_elem_list_free_space(list); +close: + snd_ctl_close(handle); + return ret; +} + +const char *snd_scenario_get_scn(struct snd_scenario *scn) +{ + if (scn->current_scenario > 0 && scn->current_scenario < MAX_SCN) + return scn->scenario[scn->current_scenario].name; + else + return NULL; +} + +int snd_scenario_set_qos(struct snd_scenario *scn, int qos) +{ + /* TODO: change QoS kcontrols */ + scn->scenario[scn->current_scenario].qos = qos; + return 0; +} + +int snd_scenario_get_qos(struct snd_scenario *scn) +{ + return scn->scenario[scn->current_scenario].qos; +} + +int snd_scenario_get_master_playback_volume(struct snd_scenario *scn) +{ + return scn->scenario[scn->current_scenario].playback_volume_id; +} + +int snd_scenario_get_master_playback_switch(struct snd_scenario *scn) +{ + return scn->scenario[scn->current_scenario].playback_switch_id; +} + +int snd_scenario_get_master_capture_volume(struct snd_scenario *scn) +{ + return scn->scenario[scn->current_scenario].capture_volume_id; +} + +int snd_scenario_get_master_capture_switch(struct snd_scenario *scn) +{ + return scn->scenario[scn->current_scenario].capture_switch_id; +} + +int snd_scenario_list(struct snd_scenario *scn, const char **list[]) +{ + *list = scn->list; + return scn->num_scenarios; +} -- 1.6.4.3 regards Stefan Schmidt

1 0

[PATCH] ascenario: Add scenario support to alsa-lib
by Stefan Schmidt 30 Sep '09

30 Sep '09

It allows switching audio settings between scenarios or uses-cases like listening to music and answering an incoming phone call. Made of control aliasing for playback, capture master and switch as well as the option to post- and prefix a sequence of control changes avoiding pops and other unwanted noise. Some example programs will be available in alsa-utils. CC: Ian Molton <ian(a)mnementh.co.uk> CC: Graeme Gregory <gg(a)slimlogic.co.uk> Signed-off-by: Liam Girdwood <lrg(a)slimlogic.co.uk> Signed-off-by: Stefan Schmidt <stefan(a)slimlogic.co.uk> --- include/Makefile.am | 2 +- include/ascenario.h | 170 +++++++ src/Makefile.am | 2 +- src/ascenario.c | 1358 +++++++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 1530 insertions(+), 2 deletions(-) create mode 100644 include/ascenario.h create mode 100644 src/ascenario.c diff --git a/include/Makefile.am b/include/Makefile.am index a291503..572fbc9 100644 --- a/include/Makefile.am +++ b/include/Makefile.am @@ -3,7 +3,7 @@ SUBDIRS = sound sysincludedir = ${includedir}/sys alsaincludedir = ${includedir}/alsa -alsainclude_HEADERS = asoundlib.h asoundef.h \ +alsainclude_HEADERS = asoundlib.h asoundef.h ascenario.h \ version.h global.h input.h output.h error.h \ conf.h control.h iatomic.h diff --git a/include/ascenario.h b/include/ascenario.h new file mode 100644 index 0000000..869f2ea --- /dev/null +++ b/include/ascenario.h @@ -0,0 +1,170 @@ +/* +* ALSA Scenario header file +* +* This library is free software; you can redistribute it and/or modify +* it under the terms of the GNU Lesser General Public License as +* published by the Free Software Foundation; either version 2.1 of +* the License, or (at your option) any later version. +* +* This program is distributed in the hope that it will be useful, +* but WITHOUT ANY WARRANTY; without even the implied warranty of +* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +* GNU Lesser General Public License for more details. +* +* You should have received a copy of the GNU Lesser General Public +* License along with this library; if not, write to the Free Software +* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA +* +* Copyright (C) 2008-2009 SlimLogic Ltd +* Authors: Liam Girdwood <lrg(a)slimlogic.co.uk> +* Stefan Schmidt <stefan(a)slimlogic.co.uk> +*/ + +/** + * Scenario IDs + * + * Standard Scenario IDs - Add new scenarios at the end. + */ + +#define SND_SCN_PLAYBACK_SPEAKER "playback speaker" +#define SND_SCN_PLAYBACK_HEADPHONES "playback headphone" +#define SND_SCN_PLAYBACK_HEADSET "playback headset" +#define SND_SCN_PLAYBACK_BLUETOOTH "playback bluetooth" +#define SND_SCN_PLAYBACK_HANDSET "playback handset" +#define SND_SCN_PLAYBACK_GSM "playback gsm" +#define SND_SCN_PLAYBACK_LINE "playback line" + +#define SND_SCN_CAPTURE_MIC "capture mic" +#define SND_SCN_CAPTURE_LINE "capture line" +#define SND_SCN_CAPTURE_HEADSET "capture headset" +#define SND_SCN_CAPTURE_HANDSET "capture handset" +#define SND_SCN_CAPTURE_BLUETOOTH "capture bluetooth" +#define SND_SCN_CAPTURE_GSM "capture gsm" + +#define SND_SCN_PHONECALL_GSM_HANDSET "phonecall gsm handset" +#define SND_SCN_PHONECALL_BT_HANDSET "phonecall bt handset" +#define SND_SCN_PHONECALL_IP_HANDSET "phonecall ip handset" +#define SND_SCN_PHONECALL_GSM_HEADSET "phonecall gsm headset" +#define SND_SCN_PHONECALL_BT_HEADSET "phonecall bt headset" +#define SND_SCN_PHONECALL_IP_HEADSET "phonecall ip headset" + +/** + * QOS + * + * Defines Audio Quality of Service. Systems supporting different types of QoS + * often have lower power consumption on lower quality levels. + */ +#define SND_POWER_QOS_HIFI 0 +#define SND_POWER_QOS_VOICE 1 +#define SND_POWER_QOS_SYSTEM 2 + +struct snd_scenario; + +/* TODO: add notification */ + +/** + * snd_scenario_list - list supported scenarios + * @scn: scenario + * @list: list of supported scenario names. + * + * List supported scenarios for this sound card. + * Returns number of scenarios. + */ +int snd_scenario_list(struct snd_scenario *scn, const char **list[]); + +/** + * snd_scenario_set_scn - set scenario + * @scn: scenario + * @scenario: scenario name + * + * Set new scenario for sound card. + */ +int snd_scenario_set_scn(struct snd_scenario *scn, const char *scenario); + +/** + * snd_scenario_get_scn - get scenario + * @scn: scenario + * + * Get current sound card scenario. + */ +const char *snd_scenario_get_scn(struct snd_scenario *scn); + +/** + * snd_scenario_get_master_playback_volume - get playback volume + * @scn: scenario + * + * Get the master playback volume control name for the current scenario. + */ +int snd_scenario_get_master_playback_volume(struct snd_scenario *scn); + +/** + * snd_scenario_get_master_playback_switch - get playback switch + * @scn: scenario + * + * Get the master playback switch control name for the current scenario. + */ + int snd_scenario_get_master_playback_switch(struct snd_scenario *scn); + +/** + * snd_scenario_get_master_capture_volume - get capture volume + * @scn: scenario + * + * Get the master capture volume control name for the current scenario. + */ +int snd_scenario_get_master_capture_volume(struct snd_scenario *scn); + +/** + * snd_scenario_get_master_capture_switch - get capture switch + * @scn: scenario + * + * Get the master capture switch control name for the current scenario. + */ +int snd_scenario_get_master_capture_switch(struct snd_scenario *scn); + +/** + * snd_scenario_set_qos - set qos + * @qos: qos + * + * Set Quality of Service for this scenario. + */ +int snd_scenario_set_qos(struct snd_scenario *scn, int qos); + +/** + * snd_scenario_get_qos - get qos + * @scn: scenario + * + * Get Quality of Service for this scenario. + */ +int snd_scenario_get_qos(struct snd_scenario *scn); + +/** + * snd_scenario_open - open scenario core + * @card_name: sound card name. + * + * Open scenario manager core for sound card. + */ +struct snd_scenario *snd_scenario_open(const char *card_name); + +/** + * snd_scenario_reload - reload and reparse scenario configuration + * @scn: scenario + * + * Reloads and reparses sound card scenario configuration. + */ +int snd_scenario_reload(struct snd_scenario *scn); + +/** + * snd_scenario_close - close scenario core + * @scn: scenario + * + * Free scenario manager core for sound card. + */ +void snd_scenario_close(struct snd_scenario *scn); + +/** + * snd_scenario_dump - dump + * @card_name: sound card name. + * + * Dump current sound card settings to stdout in scn format. + */ +int snd_scenario_dump(const char *card_name); diff --git a/src/Makefile.am b/src/Makefile.am index 3204fe4..be46cb3 100644 --- a/src/Makefile.am +++ b/src/Makefile.am @@ -14,7 +14,7 @@ SYMFUNCS = endif lib_LTLIBRARIES = libasound.la -libasound_la_SOURCES = conf.c confmisc.c input.c output.c async.c error.c dlmisc.c socket.c shmarea.c userfile.c names.c +libasound_la_SOURCES = conf.c confmisc.c input.c output.c async.c error.c dlmisc.c socket.c shmarea.c userfile.c names.c ascenario.c SUBDIRS=control libasound_la_LIBADD = control/libcontrol.la diff --git a/src/ascenario.c b/src/ascenario.c new file mode 100644 index 0000000..74082d4 --- /dev/null +++ b/src/ascenario.c @@ -0,0 +1,1358 @@ +/* + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. + * + * Copyright (C) 2008-2009 SlimLogic Ltd + * Authors: Liam Girdwood <lrg(a)slimlogic.co.uk> + * Stefan Schmidt <stefan(a)slimlogic.co.uk> + */ + +#define _GNU_SOURCE /* needed of O_NOATIME */ + +#include <stdio.h> +#include <unistd.h> +#include <errno.h> +#include <string.h> +#include <sys/types.h> +#include <sys/stat.h> +#include <fcntl.h> +#include <ctype.h> +#include <alsa/asoundlib.h> + +#include "../include/ascenario.h" + +#define PRE_SEQ 0 +#define POST_SEQ 1 +#define MAX_SCN 32 +#define MAX_NAME 64 +#define MAX_FILE 256 +#define MAX_BUF 256 +#define ALSA_SCN_DIR "/etc/alsa/scenario" + +/* + * Stores all scenario settings for 1 kcontrol. Hence we have a + * control_settings for each kcontrol in card. + */ +struct control_settings { + char name[MAX_NAME]; + int id; + snd_ctl_elem_type_t type; + int count; /* 1 = mono, 2 = stereo, etc */ + unsigned short *value; /* kcontrol value 2D array */ +}; + +/* + * If sleep is 0 the element contains the settings in control. Else sleep + * contains the sleep time in micro seconds. + */ +struct sequence_element { + unsigned int sleep; /* Sleep time in msecs if sleep element, else 0 */ + struct control_settings *control; + struct sequence_element *next; /* Pointer to next list element */ +}; + +/* + * Describes default mixers and qos for scenario. + * We have a scenario_info for each scenario loaded. + */ +struct scenario_info { + char *name; + char *file; + char *pre_sequence_file; + char *post_sequence_file; + short playback_volume_id; + short playback_switch_id; + short capture_volume_id; + short capture_switch_id; + int qos; +}; + +/* Describe a snd card and all its scenarios. + */ +struct snd_scenario { + char *card_name; + int current_scenario; + int num_scenarios; /* number of supported scenarios */ + int num_kcontrols; /* number of kcontrols */ + struct sequence_element *pre_seq_list; /* Linked list for pre sequence */ + struct sequence_element *post_seq_list; /* Linked list for post sequence */ + const char **list; + struct scenario_info *scenario; /* var len array of scenario info */ + struct control_settings *control; /* var len array of controls */ +}; + +static void scn_error(const char *fmt,...) +{ + va_list va; + va_start(va, fmt); + fprintf(stderr, "scenario: "); + vfprintf(stderr, fmt, va); + va_end(va); +} + +static void scn_stdout(const char *fmt,...) +{ + va_list va; + va_start(va, fmt); + vfprintf(stdout, fmt, va); + va_end(va); +} + +static inline void set_value(struct snd_scenario *scn, + struct control_settings *control, int count, unsigned short val) +{ + int offset = scn->current_scenario * control->count; + control->value[offset + count] = val; +} + +static inline unsigned short get_value(struct snd_scenario *scn, + struct control_settings *control, int count) +{ + int offset = scn->current_scenario * control->count; + return control->value[offset + count]; +} + +static int dump_control(snd_ctl_t *handle, snd_ctl_elem_id_t *id) +{ + int err, count, i; + snd_ctl_elem_info_t *info; + snd_ctl_elem_type_t type; + snd_ctl_elem_value_t *control; + + snd_ctl_elem_info_alloca(&info); + snd_ctl_elem_value_alloca(&control); + + snd_ctl_elem_info_set_id(info, id); + err = snd_ctl_elem_info(handle, info); + if (err < 0) { + scn_stdout("%s: failed to get ctl info\n"); + return err; + } + + snd_ctl_elem_value_set_id(control, id); + snd_ctl_elem_read(handle, control); + + type = snd_ctl_elem_info_get_type(info); + count = snd_ctl_elem_info_get_count(info); + if (count == 0) + return 0; + + scn_stdout("%u:'%s':%d:", + snd_ctl_elem_id_get_numid(id), + snd_ctl_elem_id_get_name(id), count); + + switch (type) { + case SND_CTL_ELEM_TYPE_BOOLEAN: + for (i = 0; i < count - 1; i++) + scn_stdout("%d,", + snd_ctl_elem_value_get_boolean(control, i)); + scn_stdout("%d", snd_ctl_elem_value_get_boolean(control, i)); + break; + case SND_CTL_ELEM_TYPE_INTEGER: + for (i = 0; i < count - 1; i++) + scn_stdout("%d,", + snd_ctl_elem_value_get_integer(control, i)); + scn_stdout("%d", snd_ctl_elem_value_get_integer(control, i)); + break; + case SND_CTL_ELEM_TYPE_INTEGER64: + for (i = 0; i < count - 1; i++) + scn_stdout("%ld,", + snd_ctl_elem_value_get_integer64(control, i)); + scn_stdout("%ld", + snd_ctl_elem_value_get_integer64(control, i)); + break; + case SND_CTL_ELEM_TYPE_ENUMERATED: + for (i = 0; i < count - 1; i++) + scn_stdout("%d,", + snd_ctl_elem_value_get_enumerated(control, i)); + scn_stdout("%d", + snd_ctl_elem_value_get_enumerated(control, i)); + break; + case SND_CTL_ELEM_TYPE_BYTES: + for (i = 0; i < count - 1; i++) + scn_stdout("%2.2x,", + snd_ctl_elem_value_get_byte(control, i)); + scn_stdout("%2.2x", snd_ctl_elem_value_get_byte(control, i)); + break; + default: + break; + } + scn_stdout("\n"); + return 0; +} + +/* + * Add new kcontrol from sound card into memory database. + */ +static int add_control(snd_ctl_t *handle, snd_ctl_elem_id_t *id, + struct control_settings *control_settings) +{ + int err; + snd_ctl_elem_info_t *info; + snd_ctl_elem_value_t *control; + + snd_ctl_elem_info_alloca(&info); + snd_ctl_elem_value_alloca(&control); + + snd_ctl_elem_info_set_id(info, id); + err = snd_ctl_elem_info(handle, info); + if (err < 0) { + scn_stdout("%s: failed to get ctl info\n"); + return err; + } + + snd_ctl_elem_value_set_id(control, id); + snd_ctl_elem_read(handle, control); + + strncpy(control_settings->name, snd_ctl_elem_id_get_name(id), + MAX_NAME); + control_settings->count = snd_ctl_elem_info_get_count(info); + control_settings->type = snd_ctl_elem_info_get_type(info); + control_settings->id = snd_ctl_elem_id_get_numid(id); + return 0; +} + +static int parse_controls(struct snd_scenario *scn, FILE *f) +{ + struct control_settings *control; + char buf[MAX_BUF], name[MAX_NAME]; + int id, count, line = 1, i; + char *name_start, *name_end, *tbuf; + + while (fgets(buf, MAX_BUF, f) != NULL) { + + /* get id */ + tbuf = buf; + id = atoi(tbuf); + if (id == 0) { + scn_error("%s:id == 0 on line %d\n", __func__, line); + return -EINVAL; + } + for (i = 0; i < scn->num_kcontrols; i++) { + if (id == scn->control[i].id) { + control = &scn->control[i]; + goto get_name; + } + } + scn_error("%s:id not found at line %d\n", __func__, line); + return -EINVAL; +get_name: + /* get name start */ + while (*tbuf != 0 && *tbuf != '\'') + tbuf++; + if (*tbuf == 0) + return -EINVAL; + name_start = ++tbuf; + + /* get name end */ + while (*tbuf != 0 && *tbuf != '\'') + tbuf++; + if (*tbuf == 0) + return -EINVAL; + name_end = tbuf++; + + /* copy name */ + if ((name_end - name_start) > MAX_NAME) { + scn_error("%s:name too big at %d chars line %d\n", + __func__, name_end - name_start, line); + return -EINVAL; + } + strncpy(name, name_start, name_end - name_start); + name[name_end - name_start] = 0; + if (strcmp(name, control->name)) { + scn_error("%s: name %s and %s don't match at line %d\n", + __func__, name, control->name, line); + return -EINVAL; + } + + /* get count */ + tbuf++; + count = atoi(tbuf); + if (count == 0) { + scn_error("%s:count == 0 on line %d\n", __func__, + line); + return -EINVAL; + } + if (count != control->count) { + scn_error("%s:count does not match at line %d\n", + __func__, line); + return -EINVAL; + } + + /* get vals */ + control->value = malloc(control->count * scn->num_scenarios * + sizeof(unsigned short)); + if (control->value == NULL) + return -ENOMEM; + + while (*tbuf != 0 && *tbuf != ':') + tbuf++; + if (*tbuf == 0) + return -EINVAL; + tbuf++; + + for (i = 0; i < count; i++) { + set_value(scn, control, i, atoi(tbuf)); + while (*tbuf != 0 && *tbuf != ',') + tbuf++; + + if (*tbuf++ == 0 && i < (count - 1)) + return -EINVAL; + } + line++; + } + + return 0; +} + +static char *get_string (char *buf) +{ + char *str, *end; + + /* find '=' */ + while (isblank(*buf)) + buf++; + if (*buf == 0 || *buf != '=') { + scn_error("%s: missing '='\n", __func__); + return NULL; + } + + /* find leading '"' */ + buf++; + while (isblank(*buf)) + buf++; + if (*buf == 0 || *buf != '"') { + scn_error("%s: missing start '\"'\n", __func__); + return NULL; + } + str = ++buf; + + /* get value */ + while (*buf != 0 && *buf != '"') + buf++; + end = buf; + + /* find '"' terminator */ + if (*buf == 0 || *buf != '"') { + scn_error("%s: missing terminator '\"' %s\n", __func__, buf); + return NULL; + } + + *end = 0; + return strdup(str); +} + +static char *get_control_name (char *buf) +{ + char *str, *end; + + /* find leading '"' */ + buf++; + while (isblank(*buf)) + buf++; + if (*buf == 0 || *buf != '"') { + scn_error("%s: missing start '\"'\n", __func__); + return NULL; + } + str = ++buf; + + /* get value */ + while (*buf != 0 && *buf != '"') + buf++; + end = buf; + + /* find '"' terminator */ + if (*buf == 0 || *buf != '"') { + scn_error("%s: missing terminator '\"' %s\n", __func__, buf); + return NULL; + } + + *end = 0; + return strdup(str); +} + +static int get_int (char *buf) +{ + /* find '=' */ + while (isblank(*buf)) + buf++; + if (*buf == 0 || *buf != '=') { + scn_error("%s: missing '='\n", __func__); + return -EINVAL; + } + buf++; + return atoi(buf); +} + +static int get_enum (char *buf) +{ + /* find '=' */ + while (isblank(*buf)) + buf++; + if (*buf == 0 || *buf != '=') { + scn_error("%s: missing '='\n", __func__); + return -EINVAL; + } + buf++; + return 0; /* TODO */ +} + +static void seq_list_append(struct snd_scenario *scn, + struct sequence_element *curr, int position) +{ + struct sequence_element *last, *tmp; + + if (position) { + if (!scn->post_seq_list) + scn->post_seq_list = curr; + + else { + tmp = scn->post_seq_list; + while (tmp) { + last = tmp; + tmp = tmp->next; + } + last->next = curr; + } + } + else { + if (!scn->pre_seq_list) { + scn->pre_seq_list = curr; + } + else { + tmp = scn->pre_seq_list; + while (tmp) { + last = tmp; + tmp = tmp->next; + } + last->next = curr; + } + } +} + +static int parse_sequences(struct snd_scenario *scn, FILE *f, int position) +{ + char buf[MAX_BUF], *tbuf, *control_value; + int control_len, i; + struct sequence_element *curr; + + while (fgets(buf, MAX_BUF, f) != NULL) { + + /* Check for lines with comments and ignore */ + if (buf[0] == '#') + continue; + + /* Parse current line and skip blanks */ + tbuf = buf; + while (isblank(*tbuf)) + tbuf++; + + curr = malloc(sizeof(struct sequence_element)); + if (curr == NULL) + return -ENOMEM; + bzero(curr, sizeof(struct sequence_element)); + + curr->control = malloc(sizeof(struct control_settings)); + if (curr->control == NULL) + return -ENOMEM; + bzero(curr->control, sizeof(struct control_settings)); + + curr->control->value = malloc(curr->control->count * scn->num_scenarios + * sizeof(unsigned short)); + if (curr->control->value == NULL) + return -ENOMEM; + bzero(curr->control->value, curr->control->count * scn->num_scenarios + * sizeof(unsigned short)); + + if (strncmp(tbuf, "kcontrol", 8) == 0) { + strncpy(curr->control->name, get_control_name(tbuf + 8), MAX_NAME); + control_len = strlen(curr->control->name); + /* 11 = 8 from kcontrol + 2 quotes + 1 blank */ + control_value = get_string(tbuf + 11 + control_len); + + for (i = 0; i < scn->num_kcontrols; i++) { + if (strncmp(curr->control->name, scn->control[i].name, + control_len) == 0) { + curr->sleep = 0; + curr->control->id = scn->control[i].id; + curr->control->type = scn->control[i].type; + curr->control->count = scn->control[i].count; + set_value(scn, curr->control, curr->control->count, + atoi(control_value)); + seq_list_append(scn, curr, position); + } + } + + continue; + } + + if (strncmp(tbuf, "msleep", 6) == 0) { + curr->sleep = get_int(tbuf + 6); + + /* Free control elements as we only have a sleep element + * here */ + if (curr->control) { + if (curr->control->value) + free(curr->control->value); + free(curr->control); + } + + seq_list_append(scn, curr, position); + continue; + } + } + + return 0; +} + +/* load scenario i */ +static int read_scenario_file(struct snd_scenario *scn) +{ + int fd, ret; + FILE *f; + char filename[MAX_FILE]; + struct scenario_info *info = &scn->scenario[scn->current_scenario]; + + snprintf(filename, MAX_FILE, "%s/%s/%s", ALSA_SCN_DIR, scn->card_name, + info->file); + + fd = open(filename, O_RDONLY | O_NOATIME); + if (fd < 0) { + scn_error("%s: couldn't open %s\n", __func__, filename); + return fd; + } + + f = fdopen(fd, "r"); + if (f == NULL) { + ret = errno; + goto close; + } + + ret = parse_controls(scn, f); + fclose(f); +close: + close(fd); + return ret; +} + +static int read_sequence_file(struct snd_scenario *scn, int position) +{ + int fd, ret; + FILE *f; + char filename[MAX_FILE]; + struct scenario_info *info = &scn->scenario[scn->current_scenario]; + + if (position == PRE_SEQ) { + sprintf(filename, "%s/%s/%s", ALSA_SCN_DIR, scn->card_name, + info->pre_sequence_file); + } + else { + sprintf(filename, "%s/%s/%s", ALSA_SCN_DIR, scn->card_name, + info->post_sequence_file); + } + + fd = open(filename, O_RDONLY | O_NOATIME); + if (fd < 0) { + return fd; + } + + f = fdopen(fd, "r"); + if (f == NULL) { + ret = errno; + goto close; + } + + ret = parse_sequences(scn, f, position); + fclose(f); +close: + close(fd); + return ret; +} + +static int parse_scenario(struct snd_scenario *scn, FILE *f, int line_) +{ + struct scenario_info *info; + int line = line_ - 1, id = 0, file = 0; + char buf[MAX_BUF], *tbuf; + + scn->scenario = realloc(scn->scenario, + (scn->num_scenarios + 1) * sizeof(struct scenario_info)); + if (scn->scenario == NULL) + return -ENOMEM; + bzero(scn->scenario, sizeof(struct scenario_info)); + info = scn->scenario + scn->num_scenarios; + + /* Set sequence filename to NULL as it is optional and we want to check + * for NULL to avoid segfaults */ + info->pre_sequence_file = NULL; + info->post_sequence_file = NULL; + + while(fgets(buf, MAX_BUF, f) != NULL) { + + line++; + if (buf[0] == '#') + continue; + + tbuf = buf; + while (isblank(*tbuf)) + tbuf++; + + if (strncmp(tbuf, "Identifier", 10) == 0) { + info->name = get_string(tbuf + 10); + if (info->name == NULL) { + scn_error("%s: failed to get Identifer\n", + __func__); + goto err; + } + id = 1; + continue; + } + + if (strncmp(tbuf, "File", 4) == 0) { + info->file = get_string(tbuf + 4); + if (info->file == NULL) { + scn_error("%s: failed to get File\n", + __func__); + goto err; + } + file = 1; + continue; + } + + if (strncmp(tbuf, "QoS", 3) == 0) { + info->qos = get_enum(tbuf + 3); + if (info->qos < 0) { + scn_error("%s: failed to get QoS\n", + __func__); + goto err; + } + continue; + } + + if (strncmp(tbuf, "MasterPlaybackVolume", 20) == 0) { + info->playback_volume_id = get_int(tbuf + 20); + if (info->playback_volume_id < 0) { + scn_error("%s: failed to get MasterPlaybackVolume\n", + __func__); + goto err; + } + continue; + } + + if (strncmp(tbuf, "MasterPlaybackSwitch", 20) == 0) { + info->playback_switch_id = get_int(tbuf + 20); + if (info->playback_switch_id < 0) { + scn_error("%s: failed to get MasterPlaybackSwitch\n", + __func__); + goto err; + } + continue; + } + + if (strncmp(tbuf, "MasterCaptureVolume", 19) == 0) { + info->capture_volume_id = get_int(tbuf + 19); + if (info->capture_volume_id < 0) { + scn_error("%s: failed to get MasterCaptureVolume\n", + __func__); + goto err; + } + continue; + } + + if (strncmp(tbuf, "MasterCaptureSwitch", 19) == 0) { + info->capture_switch_id = get_int(tbuf + 19); + if (info->capture_switch_id < 0) { + scn_error("%s: failed to get MasterCaptureSwitch\n", + __func__); + goto err; + } + continue; + } + + if (strncmp(tbuf, "PreSequenceFile", 15) == 0) { + info->pre_sequence_file = get_string(tbuf + 15); + if (info->pre_sequence_file == NULL) { + scn_error("%s: failed to get PreSequenceFile\n", + __func__); + goto err; + } + continue; + } + + if (strncmp(tbuf, "PostSequenceFile", 16) == 0) { + info->post_sequence_file = get_string(tbuf + 16); + if (info->post_sequence_file == NULL) { + scn_error("%s: failed to get PostSequenceFile\n", + __func__); + goto err; + } + continue; + } + + if (strncmp(tbuf, "EndSection", 10) == 0) { + break; + } + } + + if (file & id) { + scn->num_scenarios++; + return 0; + } +err: + if (file) { + free(info->file); + info->file = NULL; + } + if (id) { + free(info->name); + info->name = NULL; + } + return -EINVAL; +} + +static int read_master_file(struct snd_scenario *scn, FILE *f) +{ + int line = 0, ret = 0, i; + char buf[MAX_BUF], *tbuf; + + /* parse master config sections */ + while(fgets(buf, MAX_BUF, f) != NULL) { + + if (buf[0] == '#') { + line++; + continue; + } + + if (strncmp(buf, "Section", 7) == 0) { + + tbuf = buf + 7; + while (isblank(*tbuf)) + tbuf++; + + if (strncmp(tbuf, "\"Scenario\"", 10) == 0) { + line = parse_scenario(scn, f, line); + if (line < 0) { + scn_error("%s: failed to parse " + "scenario\n", __func__); + goto err; + } + continue; + } + } + line++; + } + + /* copy ptrs to scenario names */ + scn->list = malloc(scn->num_scenarios * sizeof(char *)); + if (scn->list == NULL) + ret = -ENOMEM; + for (i = 0; i < scn->num_scenarios; i++) + scn->list[i] = scn->scenario[i].name; + +err: + if (ferror(f)) { + scn_error("%s: failed to read master\n", __func__); + return ferror(f); + } + return ret; +} + +/* load scenario i */ +static int import_master_config(struct snd_scenario *scn) +{ + int fd, ret; + FILE *f; + char filename[MAX_FILE]; + + sprintf(filename, "%s/%s.conf", ALSA_SCN_DIR, scn->card_name); + + fd = open(filename, O_RDONLY | O_NOATIME); + if (fd < 0) { + scn_error("%s: couldn't open %s\n", __func__, filename); + return fd; + } + + f = fdopen(fd, "r"); + if (f == NULL) { + ret = errno; + goto close; + } + + ret = read_master_file(scn, f); + fclose(f); +close: + close(fd); + return ret; +} + +/* parse_card_controls + * @scn: scenario + * + * Parse sound card and store control data in memory db. + */ +static int parse_card_controls(struct snd_scenario *scn) +{ + struct control_settings *control; + snd_ctl_t *handle; + snd_ctl_card_info_t *info; + snd_ctl_elem_list_t *list; + int ret, i; + + snd_ctl_card_info_alloca(&info); + snd_ctl_elem_list_alloca(&list); + + /* open and load snd card */ + ret = snd_ctl_open(&handle, scn->card_name, SND_CTL_READONLY); + if (ret < 0) { + scn_error("%s: control %s open retor: %s\n", __func__, + scn->card_name, snd_strerror(ret)); + return ret; + } + + ret = snd_ctl_card_info(handle, info); + if (ret < 0) { + scn_error("%s :control %s local retor: %s\n", __func__, + scn->card_name, snd_strerror(ret)); + goto close; + } + + ret = snd_ctl_elem_list(handle, list); + if (ret < 0) { + scn_error("%s: cannot determine controls: %s\n", __func__, + snd_strerror(ret)); + goto close; + } + + scn->num_kcontrols = snd_ctl_elem_list_get_count(list); + if (scn->num_kcontrols < 0) { + ret = 0; + goto close; + } + + snd_ctl_elem_list_set_offset(list, 0); + if (snd_ctl_elem_list_alloc_space(list, scn->num_kcontrols) < 0) { + scn_error("%s: not enough memory...\n", __func__); + ret = -ENOMEM; + goto close; + } + if ((ret = snd_ctl_elem_list(handle, list)) < 0) { + scn_error("%s: cannot determine controls: %s\n", __func__, + snd_strerror(ret)); + goto free; + } + + /* allocate db memory for controls */ + scn->control = calloc(scn->num_kcontrols, + sizeof(struct control_settings)); + if (scn->control == NULL) { + ret = -ENOMEM; + goto free; + } + control = scn->control; + + /* iterate through each kcontrol and add to db */ + for (i = 0; i < scn->num_kcontrols; ++i) { + snd_ctl_elem_id_t *id; + snd_ctl_elem_id_alloca(&id); + snd_ctl_elem_list_get_id(list, i, id); + + ret = add_control(handle, id, control++); + if (ret < 0) { + scn_error("%s: failed to add control error %s\n", + __func__, snd_strerror(ret)); + goto close; + } + } +free: + snd_ctl_elem_list_free_space(list); +close: + snd_ctl_close(handle); + return ret; +} + +/* import_scenario_files - + * @scn: scenario + * + * Read and parse scenario_info files the store in memory. + */ +static int import_scenario_files(struct snd_scenario *scn) +{ + int ret; + + ret = import_master_config(scn); + if (ret < 0) { + scn_error("%s: failed to parse master scenario config\n", + __func__); + return ret; + } + + for (scn->current_scenario = 0; + scn->current_scenario < scn->num_scenarios; + scn->current_scenario++) { + + ret = read_scenario_file(scn); + if (ret < 0) { + scn_error("%s: failed to parse scenario %s\n", + __func__, + scn->scenario[scn->current_scenario].name); + scn->current_scenario = -1; + return ret; + } + + if (scn->scenario[scn->current_scenario].pre_sequence_file != NULL) { + ret = read_sequence_file(scn, PRE_SEQ); + if (ret < 0) { + scn_stdout("Warning: PreSequence file defined but" + " missing in scenario \"%s\"\n", + scn->scenario[scn->current_scenario].name); + } + } + + if (scn->scenario[scn->current_scenario].post_sequence_file != NULL) { + ret = read_sequence_file(scn, POST_SEQ); + if (ret < 0) { + scn_stdout("Warning: PostSequence file defined but" + " missing in scenario \"%s\"\n", + scn->scenario[scn->current_scenario].name); + } + } + + } + return 0; +} + +/* free all resorces */ +static void free_scn(struct snd_scenario *scn) +{ + /* TODO: valgrind to make sure. */ + int i; + + if (scn == NULL) + return; + + if (scn->control) { + if (scn->control->value) + free(scn->control->value); + free(scn->control); + } + + if (scn->list) + free(scn->list); + if (scn->card_name) + free(scn->card_name); + if (scn->pre_seq_list) + free(scn->pre_seq_list); + if (scn->post_seq_list) + free(scn->post_seq_list); + + if (scn->scenario) { + for (i = 0; i < scn->num_scenarios; i++) { + struct scenario_info *info = &scn->scenario[i]; + + if (info->name) + free(info->name); + if (info->file) + free(info->file); + if (info->pre_sequence_file) + free(info->pre_sequence_file); + if (info->post_sequence_file) + free(info->post_sequence_file); + } + free(scn->scenario); + } + free(scn); +} + +/* + * Init sound card scenario db. + */ +struct snd_scenario *snd_scenario_open(const char *card_name) +{ + struct snd_scenario *scn; + int err; + + /* TODO: locking and + * check if card_name scn is already loaded, + * if so reuse to conserve ram. */ + + scn = malloc(sizeof(struct snd_scenario)); + if (scn == NULL) + return NULL; + bzero(scn, sizeof(struct snd_scenario)); + scn->card_name = strdup(card_name); + if (scn->card_name == NULL) { + free(scn); + return NULL; + } + + /* get info about sound card */ + err = parse_card_controls(scn); + if (err < 0) { + free_scn(scn); + return NULL; + } + + /* get info on scenarios and verify against card */ + err = import_scenario_files(scn); + if (err < 0) { + free_scn(scn); + return NULL; + } + + return scn; +} + +/* + * Reload and reparse scenario db. + */ +int snd_scenario_reload(struct snd_scenario *scn) +{ + free_scn(scn); + + scn->num_kcontrols = parse_card_controls(scn); + if (scn->num_kcontrols <= 0) { + free_scn(scn); + return -EINVAL; + } + + scn->num_scenarios = import_scenario_files(scn); + if (scn->num_scenarios <= 0) { + return -EINVAL; + } + + return 0; +} + +void snd_scenario_close(struct snd_scenario *scn) +{ + free_scn(scn); +} + +static int set_control(snd_ctl_t *handle, snd_ctl_elem_id_t *id, + struct snd_scenario *scn) +{ + struct control_settings *setting; + int ret, count, i, idnum; + snd_ctl_elem_info_t *info; + snd_ctl_elem_type_t type; + snd_ctl_elem_value_t *control; + + snd_ctl_elem_info_alloca(&info); + snd_ctl_elem_value_alloca(&control); + + snd_ctl_elem_info_set_id(info, id); + ret = snd_ctl_elem_info(handle, info); + if (ret < 0) { + scn_error("%s: failed to get ctl info\n", __func__); + return ret; + } + + snd_ctl_elem_value_set_id(control, id); + snd_ctl_elem_read(handle, control); + + idnum = snd_ctl_elem_id_get_numid(id); + for (i = 0; i < scn->num_kcontrols; i++) { + setting = &scn->control[i]; + if (setting->id == idnum) + goto set_val; + } + scn_error("%s: failed to find control %d\n", __func__, idnum); + return 0; + +set_val: + type = snd_ctl_elem_info_get_type(info); + count = snd_ctl_elem_info_get_count(info); + if (count == 0) + return 0; + + switch (type) { + case SND_CTL_ELEM_TYPE_BOOLEAN: + for (i = 0; i < count; i++) + snd_ctl_elem_value_set_boolean(control, i, + get_value(scn, setting, i)); + break; + case SND_CTL_ELEM_TYPE_INTEGER: + for (i = 0; i < count; i++) + snd_ctl_elem_value_set_integer(control, i, + get_value(scn, setting, i)); + + break; + case SND_CTL_ELEM_TYPE_INTEGER64: + for (i = 0; i < count; i++) + snd_ctl_elem_value_set_integer64(control, i, + get_value(scn, setting, i)); + + break; + case SND_CTL_ELEM_TYPE_ENUMERATED: + for (i = 0; i < count; i++) + snd_ctl_elem_value_set_enumerated(control, i, + get_value(scn, setting, i)); + + break; + case SND_CTL_ELEM_TYPE_BYTES: + for (i = 0; i < count; i++) + snd_ctl_elem_value_set_byte(control, i, + get_value(scn, setting, i)); + break; + default: + break; + } + + ret = snd_ctl_elem_write(handle, control); + if (ret < 0) { + scn_error("%s: control %s failed: %s\n", __func__, + setting->name, snd_strerror(ret)); + scn_error("%s: count %d type: %d\n", __func__, + count, type); + for (i = 0; i < count; i++) + fprintf(stderr, "%d ", get_value(scn, setting, i)); + return ret; + } + return 0; +} + +static void exec_sequence(struct sequence_element *seq, struct snd_scenario + *scn, snd_ctl_elem_list_t *list, snd_ctl_t *handle) +{ + int count = snd_ctl_elem_list_get_count(list); + while (seq) { + if (seq->sleep) + usleep(seq->sleep); + else { + snd_ctl_elem_id_t *id; + snd_ctl_elem_id_alloca(&id); + int ret, i, numid; + /* Where is id lookup from numid if you need it? */ + for (i = 0; i < count; ++i) { + snd_ctl_elem_list_get_id(list, i, id); + numid = snd_ctl_elem_id_get_numid(id); + if (numid == seq->control->id) { + ret = set_control(handle, id, scn); + if (ret < 0) { + scn_error("%s: failed to set control %s\n", + __func__, scn->card_name); + } + break; + } + } + } + seq = seq->next; + } +} + +int snd_scenario_set_scn(struct snd_scenario *scn, const char *name) +{ + snd_ctl_card_info_t *info; + snd_ctl_elem_list_t *list; + snd_ctl_t *handle; + int ret, count, i; + + snd_ctl_card_info_alloca(&info); + snd_ctl_elem_list_alloca(&list); + + /* find scenario name */ + for (i = 0; i < scn->num_scenarios; i++) { + if (!strcmp(scn->scenario[i].name, name)) + goto found; + } + scn_error("%s: scenario %s not found\n", __func__, name); + return -EINVAL; + +found: + /* scenario found - now open card */ + scn->current_scenario = i; + ret = snd_ctl_open(&handle, scn->card_name, 0); + if (ret) { + scn_error("%s: control %s open error: %s\n", __func__, + scn->card_name, snd_strerror(ret)); + return ret; + } + + ret = snd_ctl_card_info(handle, info); + if (ret < 0) { + scn_error("%s :control %s local retor: %s\n", __func__, + scn->card_name, snd_strerror(ret)); + goto close; + } + + ret = snd_ctl_elem_list(handle, list); + if (ret < 0) { + scn_error("%s: cannot determine controls: %s\n", __func__, + snd_strerror(ret)); + goto close; + } + + count = snd_ctl_elem_list_get_count(list); + if (count < 0) { + ret = 0; + goto close; + } + + snd_ctl_elem_list_set_offset(list, 0); + if (snd_ctl_elem_list_alloc_space(list, count) < 0) { + scn_error("%s: not enough memory...\n", __func__); + ret = -ENOMEM; + goto close; + } + if ((ret = snd_ctl_elem_list(handle, list)) < 0) { + scn_error("%s: cannot determine controls: %s\n", __func__, + snd_strerror(ret)); + goto free; + } + + /* If we have a sequence list that should be executed before the new + * scenario is set do it now */ + if (scn->pre_seq_list) + exec_sequence(scn->pre_seq_list, scn, list, handle); + + /* iterate through each kcontrol and add to db */ + for (i = 0; i < count; ++i) { + snd_ctl_elem_id_t *id; + snd_ctl_elem_id_alloca(&id); + snd_ctl_elem_list_get_id(list, i, id); + + ret = set_control(handle, id, scn); + if (ret < 0) { + scn_error("%s: failed to set control %s\n", __func__, + scn->card_name); + } + } + + /* If we have a sequence list that should be executed after the new + * scenario is set do it now */ + if (scn->post_seq_list) + exec_sequence(scn->post_seq_list, scn, list, handle); + +free: + snd_ctl_elem_list_free_space(list); +close: + snd_ctl_close(handle); + return ret; +} + +int snd_scenario_dump(const char *card_name) +{ + snd_ctl_t *handle; + snd_ctl_card_info_t *info; + snd_ctl_elem_list_t *list; + int ret, i, count; + + snd_ctl_card_info_alloca(&info); + snd_ctl_elem_list_alloca(&list); + + /* open and load snd card */ + ret = snd_ctl_open(&handle, card_name, SND_CTL_READONLY); + if (ret < 0) { + scn_error("%s: control %s open retor: %s\n", __func__, card_name, + snd_strerror(ret)); + return ret; + } + + ret = snd_ctl_card_info(handle, info); + if (ret < 0) { + scn_error("%s :control %s local retor: %s\n", __func__, + card_name, snd_strerror(ret)); + goto close; + } + + ret = snd_ctl_elem_list(handle, list); + if (ret < 0) { + scn_error("%s: cannot determine controls: %s\n", __func__, + snd_strerror(ret)); + goto close; + } + + count = snd_ctl_elem_list_get_count(list); + if (count < 0) { + ret = 0; + goto close; + } + + snd_ctl_elem_list_set_offset(list, 0); + if (snd_ctl_elem_list_alloc_space(list, count) < 0) { + scn_error("%s: not enough memory...\n", __func__); + ret = -ENOMEM; + goto close; + } + if ((ret = snd_ctl_elem_list(handle, list)) < 0) { + scn_error("%s: cannot determine controls: %s\n", __func__, + snd_strerror(ret)); + goto free; + } + + /* iterate through each kcontrol and add to db */ + for (i = 0; i < count; ++i) { + snd_ctl_elem_id_t *id; + snd_ctl_elem_id_alloca(&id); + snd_ctl_elem_list_get_id(list, i, id); + + ret = dump_control(handle, id); + if (ret < 0) { + scn_error("%s: cannot determine controls: %s\n", + __func__, snd_strerror(ret)); + goto free; + } + } +free: + snd_ctl_elem_list_free_space(list); +close: + snd_ctl_close(handle); + return ret; +} + +const char *snd_scenario_get_scn(struct snd_scenario *scn) +{ + if (scn->current_scenario > 0 && scn->current_scenario < MAX_SCN) + return scn->scenario[scn->current_scenario].name; + else + return NULL; +} + +int snd_scenario_set_qos(struct snd_scenario *scn, int qos) +{ + /* TODO: change QoS kcontrols */ + scn->scenario[scn->current_scenario].qos = qos; + return 0; +} + +int snd_scenario_get_qos(struct snd_scenario *scn) +{ + return scn->scenario[scn->current_scenario].qos; +} + +int snd_scenario_get_master_playback_volume(struct snd_scenario *scn) +{ + return scn->scenario[scn->current_scenario].playback_volume_id; +} + +int snd_scenario_get_master_playback_switch(struct snd_scenario *scn) +{ + return scn->scenario[scn->current_scenario].playback_switch_id; +} + +int snd_scenario_get_master_capture_volume(struct snd_scenario *scn) +{ + return scn->scenario[scn->current_scenario].capture_volume_id; +} + +int snd_scenario_get_master_capture_switch(struct snd_scenario *scn) +{ + return scn->scenario[scn->current_scenario].capture_switch_id; +} + +int snd_scenario_list(struct snd_scenario *scn, const char **list[]) +{ + *list = scn->list; + return scn->num_scenarios; +} -- 1.6.4.3 regards Stefan Schmidt

1 0

[alsa-devel] Some bugs with Auzentech Prelude
by Laurent Léonard 30 Sep '09

30 Sep '09

Hi, I just bought an Auzentech Prelude and compiled a 2.6.31.1 kernel to test it. Analog and digital outputs are working but I experience some problems: - There are some lags when I play long tracks on Amarok. - Sometimes the track suddenly play slower for a time and go back, or not, and I have to pause and play to fix it. - Sometimes there is noise (only) for a time when I play a track, then it go back after a while. - I was unable to play an audio file correctly with VLC, the sound is very saccaded. - No sound at all with Flash and Iceweasel. Are they known bugs ? Are there plans to fix them ? Thank you, -- Laurent Léonard

1 0

[alsa-devel] [PATCH 0/3] Usage of AM_SILENT_RULES and small cleanup
by Stefan Schmidt 30 Sep '09

30 Sep '09

Hello. While working with alsa-lib to get scenarios in shape and submitted later I added support for the new automake feature AM_SILENT_RULES and did some small cleanups. [PATCH 1/3] cvscompile: Remove in favour of gitcompile. I see no reason to keep this around when CVS is no longer used and gitcompile does replace it. Something I miss here? [PATCH 2/3] configure.in: Add m4 check for new AM_SILENT_RULES The kbuild like output makes it a lot easier to follow the compile and linking process, at least in my opinion. It is available with automake 1.11 and we use a m4 macro here to only use it when it is available. [PATCH 3/3] control: Remove unused variable. Found while really seeing what is going on thanks to AM_SILENT_RULES: ;) There are also some unused paramter and a function around. I don't know how you guys prefer to handle this in the ALSA project so I did not prepare patches for this yet. Let me know if you want some for it. pcm_route.c: In function ‘snd_pcm_route_convert1_many’: pcm_route.c:350: warning: ‘src’ may be used uninitialized in this function pcm_rate_linear.c: In function ‘get_supported_rates’: pcm_rate_linear.c:408: warning: unused parameter ‘rate’ pcm_rate_linear.c: In function ‘linear_dump’: pcm_rate_linear.c:416: warning: unused parameter ‘rate’ pcm_rate_linear.c: In function ‘_snd_pcm_rate_linear_open’: pcm_rate_linear.c:435: warning: unused parameter ‘version’ alisp.c:237: warning: ‘incref_tree_explicit’ defined but not used regards Stefan Schmidt

3 7

[alsa-devel] [PATCH 1/2] pcm_rate_linear: Annotate unused function parameter to avoid compiler warnings.
by Stefan Schmidt 30 Sep '09

30 Sep '09

From: Stefan Schmidt <stefan(a)slimlogic.co.uk> Signed-off-by: Stefan Schmidt <stefan(a)slimlogic.co.uk> --- src/pcm/pcm_rate_linear.c | 9 +++++---- 1 files changed, 5 insertions(+), 4 deletions(-) diff --git a/src/pcm/pcm_rate_linear.c b/src/pcm/pcm_rate_linear.c index 8b2d2d0..7481b38 100644 --- a/src/pcm/pcm_rate_linear.c +++ b/src/pcm/pcm_rate_linear.c @@ -405,15 +405,15 @@ static void linear_close(void *obj) free(obj); } -static int get_supported_rates(void *rate, unsigned int *rate_min, - unsigned int *rate_max) +static int get_supported_rates(ATTRIBUTE_UNUSED void *rate, + unsigned int *rate_min, unsigned int *rate_max) { *rate_min = SND_PCM_PLUGIN_RATE_MIN; *rate_max = SND_PCM_PLUGIN_RATE_MAX; return 0; } -static void linear_dump(void *rate, snd_output_t *out) +static void linear_dump(ATTRIBUTE_UNUSED void *rate, snd_output_t *out) { snd_output_printf(out, "Converter: linear-interpolation\n"); } @@ -432,7 +432,8 @@ static const snd_pcm_rate_ops_t linear_ops = { .dump = linear_dump, }; -int SND_PCM_RATE_PLUGIN_ENTRY(linear) (unsigned int version, void **objp, snd_pcm_rate_ops_t *ops) +int SND_PCM_RATE_PLUGIN_ENTRY(linear) (ATTRIBUTE_UNUSED unsigned int version, + void **objp, snd_pcm_rate_ops_t *ops) { struct rate_linear *rate; -- 1.6.4.3

1 1

[alsa-devel] [PATCH 0/1] ASoC: DAPM: Fix for line widget handling
by Peter Ujfalusi 30 Sep '09

30 Sep '09

Hello, The following patch fixes the handling of the line widget, which can be input or output. Additional check is added to is_connected_output_ep and is_connected_input_ep functions when the line widget is checked to handle only the appropriate line widget. Without the patch these two functions would pick up either line widgets, which confused DAPM. Note: checkpach complains for the long lines, but I feel that it is appropriate here to use the long lines. --- Peter Ujfalusi (1): ASoC: Fix SND_SOC_DAPM_LINE handling sound/soc/soc-dapm.c | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-)

2 2

[alsa-devel] [PATCH 1/1] Echoaudio: fix broken Mia driver
by Giuliano Pochini 30 Sep '09

30 Sep '09

This change: http://git.alsa-project.org/?p=alsa-kernel.git;a=blobdiff;f=sound/pci/echoa… broke the Mia driver because it removed the line-out control interface from all vmixer cards. But the Mia has that (undocumented!!) control and after the change it is impossible to raise the volume. I tested the driver with my IndigoIOx that was supposed to have the same set of controls... sorry. I hope it's not too late to merge this patch in linux-2.6.32. ---------- Re-enable the line-out control for the Mia card. Signed-off-by: Giuliano Pochini <pochini(a)shiny.it> diff -u alsa-driver-1.0.21/alsa-kernel/pci/echoaudio__orig/echoaudio.c alsa-driver-1.0.21/alsa-kernel/pci/echoaudio/echoaudio.c --- alsa-driver-1.0.21/alsa-kernel/pci/echoaudio__orig/echoaudio.c 2009-08-28 13:41:39.000000000 +0200 +++ alsa-driver-1.0.21/alsa-kernel/pci/echoaudio/echoaudio.c 2009-09-29 21:55:48.000000000 +0200 @@ -950,7 +950,7 @@ Control interface ******************************************************************************/ -#ifndef ECHOCARD_HAS_VMIXER +#if !defined(ECHOCARD_HAS_VMIXER) || defined(ECHOCARD_HAS_LINE_OUT_GAIN) /******************* PCM output volume *******************/ static int snd_echo_output_gain_info(struct snd_kcontrol *kcontrol, @@ -1003,6 +1003,19 @@ return changed; } +#ifdef ECHOCARD_HAS_LINE_OUT_GAIN +/* On the Mia this one controls the line-out volume */ +static struct snd_kcontrol_new snd_echo_line_output_gain __devinitdata = { + .name = "Line Playback Volume", + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .info = snd_echo_output_gain_info, + .get = snd_echo_output_gain_get, + .put = snd_echo_output_gain_put, + .tlv = {.p = db_scale_output_gain}, +}; +#else static struct snd_kcontrol_new snd_echo_pcm_output_gain __devinitdata = { .name = "PCM Playback Volume", .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1012,9 +1025,10 @@ .put = snd_echo_output_gain_put, .tlv = {.p = db_scale_output_gain}, }; - #endif +#endif /* !ECHOCARD_HAS_VMIXER || ECHOCARD_HAS_LINE_OUT_GAIN */ + #ifdef ECHOCARD_HAS_INPUT_GAIN @@ -2030,10 +2044,14 @@ snd_echo_vmixer.count = num_pipes_out(chip) * num_busses_out(chip); if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_vmixer, chip))) < 0) goto ctl_error; -#else - if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_pcm_output_gain, chip))) < 0) +#ifdef ECHOCARD_HAS_LINE_OUT_GAIN + if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_line_output_gain, chip))) < 0) goto ctl_error; #endif +#else /* ECHOCARD_HAS_VMIXER */ + if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_pcm_output_gain, chip))) < 0) + goto ctl_error; +#endif /* ECHOCARD_HAS_VMIXER */ #ifdef ECHOCARD_HAS_INPUT_GAIN if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_line_input_gain, chip))) < 0) diff -u alsa-driver-1.0.21/alsa-kernel/pci/echoaudio__orig/mia.c alsa-driver-1.0.21/alsa-kernel/pci/echoaudio/mia.c --- alsa-driver-1.0.21/alsa-kernel/pci/echoaudio__orig/mia.c 2009-08-28 13:41:39.000000000 +0200 +++ alsa-driver-1.0.21/alsa-kernel/pci/echoaudio/mia.c 2009-09-29 21:55:48.000000000 +0200 @@ -29,6 +29,7 @@ #define ECHOCARD_HAS_ADAT FALSE #define ECHOCARD_HAS_STEREO_BIG_ENDIAN32 #define ECHOCARD_HAS_MIDI +#define ECHOCARD_HAS_LINE_OUT_GAIN /* Pipe indexes */ #define PX_ANALOG_OUT 0 /* 8 */ -- Giuliano.

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[alsa-devel] alsa duplex test application
by 罗富冲 30 Sep '09

30 Sep '09

Hi all, I want to write a sound card Duplex test Application(playing what we are recording) refer to alsa-utils, and i got Segmentation fault. Accessoriesis my application for test,can anyone help me to find out what is the problem, or show me a application which can work. Sorry for my poor english, and waiting for your reply.

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[alsa-devel] My Codec: IDT 92HD75B3X5 issues
by Munzir Taha (منذر طه) 29 Sep '09

29 Sep '09

Hi sirs, These are the issues I have with my Laptop HP Pavilion dv6 1225ee regarding audio support. 1. No sense for headphone jack (jack_present?), when headphone is plugged (present_int_hp?) 2. There headphone mic jack is not working. The one next to the webcam is always recording even when I change the "Mic Jack" control from "Mic In" to "Line In" 3. The mute/unmute LED doesn't change color to RED when its muted like in Windows. Touching that LED shows this in /var/log/messages: Sep 28 06:53:47 dv6 kernel: [ 9197.986789] atkbd.c: Unknown key released (translated set 2, code 0x94 on isa0060/serio0). Sep 28 06:53:47 dv6 kernel: [ 9197.986791] atkbd.c: Use 'setkeycodes e014 <keycode>' to make it known. Sep 28 06:53:47 dv6 kernel: [ 9198.353568] atkbd.c: Unknown key pressed (translated set 2, code 0x94 on isa0060/serio0). Sep 28 06:53:47 dv6 kernel: [ 9198.353572] atkbd.c: Use 'setkeycodes e014 <keycode>' to make it known. Sep 28 06:53:47 dv6 kernel: [ 9198.359205] atkbd.c: Unknown key released (translated set 2, code 0x94 on isa0060/serio0). Sep 28 06:53:47 dv6 kernel: [ 9198.359208] atkbd.c: Use 'setkeycodes e014 <keycode>' to make it known. alsa info is at: http://www.alsa-project.org/db/?f=8f8518c634bc6880a1633e025fe7a7f975f3ed28 I also played with different models like hp-m4 HP mini 1000 hp-dv5 HP dv series hp-hdx HP HDX series hp-dv4-1222nr HP dv4-1222nr (with LED support) but nothing helps. -- Munzir Taha Telecommunications and Electronics Engineer Master CIW Designer, ICDL, MOUS, Linux+, LPI, SCSA Riyadh, SA

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[alsa-devel] ASoC: SND_SOC_DAPM_LINE behavior
by Peter Ujfalusi 29 Sep '09

29 Sep '09

Hello, as I'm writing a new codec driver I have noticed the following (using 2.6.31 kernel). The codec itself is really simple, it is only capable of playback and it can bypass audio from line-in to the output. The DAPM routing is something like this in the codec driver: |DAC|------------->| | |Playback Mixer|->|OUTPUT| |INPUT|->|Bypass|->| | |SWITCH| Now if in the machine driver I create the following DAPM widget: SND_SOC_DAPM_LINE("Line In", NULL), and than connect this DAPM_LINE to the codec's INPUT (LINE-IN): {"LINE-IN", NULL, "Line In"}, Than the codec bias level would be always in ON state, regardless of the state of the Bypass Switch (it is off by default). I can use the snd_soc_dapm_disable_pin(codec, "Line In") in the machine driver to disable the pin, which moves the codec to STANDBY state. Now, if I change the SND_SOC_DAPM_LINE to SND_SOC_DAPM_MIC("Line In", NULL), in the machine driver, than the codec bias would be STANDBY as long as the Bypass switches are off, if they changed to on, than the codec bias goes to ON, as expected. Is this expected behavior? -- Péter

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[alsa-devel] hda-intel: low volume with nvidia mcp7a hdmi
by Halim Sahin 29 Sep '09

29 Sep '09

Hello, The volume of the mentioned card is really too low. used kernel: 2.6.31-rc6 from opensuse factory. lspci reports: 00:08.0 Audio device: nVidia Corporation MCP79 High Definition Audio (rev b1) Subsystem: nVidia Corporation Device cb84 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Ste Status: Cap+ 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- <TAbort- Latency: 0 (500ns min, 1250ns max) Interrupt: pin A routed to IRQ 21 Region 0: Memory at fad78000 (32-bit, non-prefetchable) [size=16K] Capabilities: [44] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3ho Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: HDA Intel Output from /proc/asound/cards: 0 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xfad78000 irq 21 I didn't give any cmdline params to snd-ihda-intel. Any ideas how to get more power out of this card? regards halim

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[alsa-devel] [PATCH] ASoC: Consider the codec->active also when changing bias level
by Peter Ujfalusi 29 Sep '09

29 Sep '09

Take into account the codec->active (playback/capture activity) when changing the bias level. Fixes the following scenario: if during active playback/capture the DAPM routes has been modified in a way, that there will be no complete route (for example muting the output) DAPM would set the codec into BIAS_STANDBY, which could disable the clocks from the codec (codec is master), which breaks the playback with error since the clocks are missing from the serial bus and no data is shifted out/in. With this patch during active use (playback/capture) the DAPM will not set the bias to STANDBY, thus keeping the playback/capture running. When no playback/capture is running, the DAPM will handle the complete paths in a same way it used to: setting the bias to ON/STANDBY based on complete paths. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.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 9babda5..567be82 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -977,7 +977,7 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) continue; power = w->power_check(w); - if (power) + if (power || codec->active) sys_power = 1; if (w->power == power) -- 1.6.5.rc1

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[alsa-devel] Alsa not working when linked to a library
by Fortini Matteo 29 Sep '09

29 Sep '09

Hi, I'm experiencing a very strange behavior. I'm using alsa 1.0.18 on a powerpc platform. If I run the following simple code from the main of a .c file, it works perfectly, but if I put into a shared .so library (libaudio.so), then I link the main with -laudio -lasound, then the hwparams structure is not filled with anything, while the alsa functions don't return any error. Do you have any clue why this could happen? TIA ############################################################ int play_sample(char *pcm_name, char *samplename) { int res; snd_pcm_t *pcm_handle; snd_pcm_hw_params_t *hwparams; const char *name= (pcm_name != NULL) ? pcm_name : "speaker"; if ( (res = snd_pcm_open(&pcm_handle, name, SND_PCM_STREAM_PLAYBACK, 0)) < 0) { fprintf(stderr, "Error opening PCM device %s\n", pcm_name); return 1; } else { printf ("snd_pcm_open returned %d\n", res); } snd_pcm_hw_params_malloc(&hwparams); if (snd_pcm_hw_params_any(pcm_handle, hwparams) < 0) { fprintf(stderr, "Can not configure this PCM device.\n"); return 1; snd_pcm_close(pcm_handle); } int main (void) { play_sample (NULL, "test_sound.wav); return 0; } ############################################################

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Re: [alsa-devel] WM8731 using I2S on omap3 McBSP2 issues
by Rick Bronson 29 Sep '09

29 Sep '09

Peter, I'm getting a XUNDFLSTAT (indicates a real underflow condition) and XEMPTYEOF (Transmit buffer empty at end of frame) in MCBSPLP_IRQSTATUS_REG. Does this tell you anything? Rick

2 1

[alsa-devel] [PATCH 1/1] ALSA: hda - Analog Devices AD1984A add HP Touchsmart model
by Miguel de Barros 29 Sep '09

29 Sep '09

Hello, I would like to submit this patch, it will fix the bug I've submited : https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4614 Thanks, Miguel de Barros

2 3

[alsa-devel] [PATCH] snd-usb-us122l: added support for US-144
by Eric Luiggi 29 Sep '09

29 Sep '09

Hi Tobias, I'd like to get my US-144 interface working on jaunty and am willing to help out with testing for this US-144 solution. Please send patch instructions. Thanks, Eric

2 1

Re: [alsa-devel] [PATCH] sscape: convert to firmware loader framework
by krzysztof.h1＠poczta.fm 29 Sep '09

29 Sep '09

Please drop this patch. It does not compile because it includes removed sscape_ioctl.h. I will send updated patch. Best regards, Krzysztof "Krzysztof Helt" <krzysztof.h1(a)poczta.fm> pisze: > The conversion solves the problem that firmware size was set to 64KB > while non PnP cards have 128KB firmware files. > > An additional post firmware initialization code has been moved > from the OSS driver. > > --- > Now, the MIDI is audible but not correct. It is not played timely. > The sscape_ctl alsa-tool is not needed any more. > > > Documentation/sound/alsa/ALSA-Configuration.txt | 8 +- > include/sound/sscape_ioctl.h | 21 -- > sound/isa/Kconfig | 8 +- > sound/isa/sscape.c | 305 > +++++++---------------- > 4 files changed, 101 insertions(+), 241 deletions(-) > delete mode 100644 include/sound/sscape_ioctl.h > > diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt > b/Documentation/sound/alsa/ALSA-Configuration.txt > index 1c8eb45..cf98525 100644 > --- a/Documentation/sound/alsa/ALSA-Configuration.txt > +++ b/Documentation/sound/alsa/ALSA-Configuration.txt > @@ -1631,7 +1631,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' > prefix. This was removed. > Module snd-sscape > ----------------- > > - Module for ENSONIQ SoundScape PnP cards. > + Module for ENSONIQ SoundScape cards. > > port - Port # (PnP setup) > wss_port - WSS Port # (PnP setup) > @@ -1640,9 +1640,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' > prefix. This was removed. > dma - DMA # (PnP setup) > dma2 - 2nd DMA # (PnP setup, -1 to disable) > > - This module supports multiple cards. ISA PnP must be enabled. > - You need sscape_ctl tool in alsa-tools package for loading > - the microcode. > + This module supports multiple cards. > + > + The driver requires the firmware loader support on kernel. > > Module snd-sun-amd7930 (on sparc only) > -------------------------------------- > diff --git a/include/sound/sscape_ioctl.h b/include/sound/sscape_ioctl.h > deleted file mode 100644 > index 0d88859..0000000 > --- a/include/sound/sscape_ioctl.h > +++ /dev/null > @@ -1,21 +0,0 @@ > -#ifndef SSCAPE_IOCTL_H > -#define SSCAPE_IOCTL_H > - > - > -struct sscape_bootblock > -{ > - unsigned char code[256]; > - unsigned version; > -}; > - > -#define SSCAPE_MICROCODE_SIZE 65536 > - > -struct sscape_microcode > -{ > - unsigned char __user *code; > -}; > - > -#define SND_SSCAPE_LOAD_BOOTB _IOWR('P', 100, struct sscape_bootblock) > -#define SND_SSCAPE_LOAD_MCODE _IOW ('P', 101, struct sscape_microcode) > - > -#endif > diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig > index b90fc16..02fe81c 100644 > --- a/sound/isa/Kconfig > +++ b/sound/isa/Kconfig > @@ -372,9 +372,9 @@ config SND_SGALAXY > > config SND_SSCAPE > tristate "Ensoniq SoundScape driver" > - select SND_HWDEP > select SND_MPU401_UART > select SND_WSS_LIB > + select FW_LOADER > help > Say Y here to include support for Ensoniq SoundScape > and Ensoniq OEM soundcards. > @@ -382,7 +382,11 @@ config SND_SSCAPE > The PCM audio is supported on SoundScape Classic, Elite, PnP > and VIVO cards. The supported OEM cards are SPEA Media FX and > Reveal SC-600. > - The MIDI support is very experimental. > + The MIDI support is very experimental and requires binary > + firmware files called "scope.cod" and "sndscape.co?" where the > + ? is digit 0, 1, 2, 3 or 4. The firmware files can be found > + in DOS or Windows driver packages. One has to put the firmware > + files into the /lib/firmware directory. > > To compile this driver as a module, choose M here: the module > will be called snd-sscape. > diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c > index b11c35f..08389a2 100644 > --- a/sound/isa/sscape.c > +++ b/sound/isa/sscape.c > @@ -1,5 +1,5 @@ > /* > - * Low-level ALSA driver for the ENSONIQ SoundScape PnP > + * Low-level ALSA driver for the ENSONIQ SoundScape > * Copyright (c) by Chris Rankin > * > * This driver was written in part using information obtained from > @@ -25,12 +25,12 @@ > #include <linux/err.h> > #include <linux/isa.h> > #include <linux/delay.h> > +#include <linux/firmware.h> > #include <linux/pnp.h> > #include <linux/spinlock.h> > #include <linux/moduleparam.h> > #include <asm/dma.h> > #include <sound/core.h> > -#include <sound/hwdep.h> > #include <sound/wss.h> > #include <sound/mpu401.h> > #include <sound/initval.h> > @@ -39,7 +39,7 @@ > > > MODULE_AUTHOR("Chris Rankin"); > -MODULE_DESCRIPTION("ENSONIQ SoundScape PnP driver"); > +MODULE_DESCRIPTION("ENSONIQ SoundScape driver"); > MODULE_LICENSE("GPL"); > > static int index[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IDX; > @@ -142,14 +142,12 @@ struct soundscape { > struct resource *wss_res; > struct snd_wss *chip; > struct snd_mpu401 *mpu; > - struct snd_hwdep *hw; > > /* > * The MIDI device won't work until we've loaded > * its firmware via a hardware-dependent device IOCTL > */ > spinlock_t fwlock; > - int hw_in_use; > unsigned long midi_usage; > unsigned char midi_vol; > }; > @@ -167,12 +165,6 @@ static inline struct soundscape > *get_mpu401_soundscape(struct snd_mpu401 * mpu) > return (struct soundscape *) (mpu->private_data); > } > > -static inline struct soundscape *get_hwdep_soundscape(struct snd_hwdep * > hw) > -{ > - return (struct soundscape *) (hw->private_data); > -} > - > - > /* > * Allocates some kernel memory that we can use for DMA. > * I think this means that the memory has to map to > @@ -438,14 +430,14 @@ static int host_startup_ack(struct soundscape *s, > unsigned timeout) > * Upload a byte-stream into the SoundScape using DMA channel A. > */ > static int upload_dma_data(struct soundscape *s, > - const unsigned char __user *data, > + const unsigned char *data, > size_t size) > { > unsigned long flags; > struct snd_dma_buffer dma; > int ret; > > - if (!get_dmabuf(&dma, PAGE_ALIGN(size))) > + if (!get_dmabuf(&dma, PAGE_ALIGN(32 * 1024))) > return -ENOMEM; > > spin_lock_irqsave(&s->lock, flags); > @@ -468,35 +460,17 @@ static int upload_dma_data(struct soundscape *s, > sscape_write_unsafe(s->io_base, GA_HMCTL_REG, > sscape_read_unsafe(s->io_base, GA_HMCTL_REG) | 0x80); > > /* > - * Upload the user's data (firmware?) to the SoundScape > + * Upload the firmware to the SoundScape > * board through the DMA channel ... > */ > while (size != 0) { > unsigned long len; > > - /* > - * Apparently, copying to/from userspace can sleep. > - * We are therefore forbidden from holding any > - * spinlocks while we copy ... > - */ > - spin_unlock_irqrestore(&s->lock, flags); > - > - /* > - * Remember that the data that we want to DMA > - * comes from USERSPACE. We have already verified > - * the userspace pointer ... > - */ > len = min(size, dma.bytes); > - len -= __copy_from_user(dma.area, data, len); > + memcpy(dma.area, data, len); > data += len; > size -= len; > > - /* > - * Grab that spinlock again, now that we've > - * finished copying! > - */ > - spin_lock_irqsave(&s->lock, flags); > - > snd_dma_program(s->chip->dma1, dma.addr, len, DMA_MODE_WRITE); > sscape_start_dma_unsafe(s->io_base, GA_DMAA_REG); > if (!sscape_wait_dma_unsafe(s->io_base, GA_DMAA_REG, 5000)) { > @@ -512,6 +486,7 @@ static int upload_dma_data(struct soundscape *s, > } /* while */ > > set_host_mode_unsafe(s->io_base); > + outb(0x0, s->io_base); > > /* > * Boot the board ... (I think) > @@ -537,7 +512,7 @@ _release_dma: > /* > * NOTE!!! We are NOT holding any spinlocks at this point !!! > */ > - sscape_write(s, GA_DMAA_REG, (s->ic_type == IC_ODIE ? 0x70 : 0x40)); > + sscape_write(s, GA_DMAA_REG, (s->ic_type == IC_OPUS ? 0x40 : 0x70)); > free_dmabuf(&dma); > > return ret; > @@ -547,162 +522,70 @@ _release_dma: > * Upload the bootblock(?) into the SoundScape. The only > * purpose of this block of code seems to be to tell > * us which version of the microcode we should be using. > - * > - * NOTE: The boot-block data resides in USER-SPACE!!! > - * However, we have already verified its memory > - * addresses by the time we get here. > */ > -static int sscape_upload_bootblock(struct soundscape *sscape, struct > sscape_bootblock __user *bb) > +static int sscape_upload_bootblock(struct snd_card *card) > { > + struct soundscape *sscape = get_card_soundscape(card); > unsigned long flags; > + const struct firmware *init_fw = NULL; > int data = 0; > int ret; > > - ret = upload_dma_data(sscape, bb->code, sizeof(bb->code)); > - > - spin_lock_irqsave(&sscape->lock, flags); > - if (ret == 0) { > - data = host_read_ctrl_unsafe(sscape->io_base, 100); > - } > - set_midi_mode_unsafe(sscape->io_base); > - spin_unlock_irqrestore(&sscape->lock, flags); > - > - if (ret == 0) { > - if (data < 0) { > - snd_printk(KERN_ERR "sscape: timeout reading firmware versionn"); > - ret = -EAGAIN; > - } > - else if (__copy_to_user(&bb->version, &data, sizeof(bb->version))) { > - ret = -EFAULT; > - } > + ret = request_firmware(&init_fw, "scope.cod", card->dev); > + if (ret < 0) { > + snd_printk(KERN_ERR "Error loading scope.cod"); > + return ret; > } > + ret = upload_dma_data(sscape, init_fw->data, init_fw->size); > > - return ret; > -} > + release_firmware(init_fw); > > -/* > - * Upload the microcode into the SoundScape. The > - * microcode is 64K of data, and if we try to copy > - * it into a local variable then we will SMASH THE > - * KERNEL'S STACK! We therefore leave it in USER > - * SPACE, and save ourselves from copying it at all. > - */ > -static int sscape_upload_microcode(struct soundscape *sscape, > - const struct sscape_microcode __user > *mc) > -{ > - unsigned long flags; > - char __user *code; > - int err; > - > - /* > - * We are going to have to copy this data into a special > - * DMA-able buffer before we can upload it. We shall therefore > - * just check that the data pointer is valid for now. > - * > - * NOTE: This buffer is 64K long! That's WAY too big to > - * copy into a stack-temporary anyway. > - */ > - if ( get_user(code, &mc->code) || > - !access_ok(VERIFY_READ, code, SSCAPE_MICROCODE_SIZE) ) > - return -EFAULT; > + spin_lock_irqsave(&sscape->lock, flags); > + if (ret == 0) > + data = host_read_ctrl_unsafe(sscape->io_base, 100); > > - if ((err = upload_dma_data(sscape, code, SSCAPE_MICROCODE_SIZE)) == 0) > { > - snd_printk(KERN_INFO "sscape: MIDI firmware loadedn"); > - } > + if (data & 0x10) > + sscape_write_unsafe(sscape->io_base, GA_SMCFGA_REG, 0x2f); > > - spin_lock_irqsave(&sscape->lock, flags); > - set_midi_mode_unsafe(sscape->io_base); > spin_unlock_irqrestore(&sscape->lock, flags); > > - initialise_mpu401(sscape->mpu); > + data &= 0xf; > + if (ret == 0 && data > 7) { > + snd_printk(KERN_ERR > + "sscape: timeout reading firmware versionn"); > + ret = -EAGAIN; > + } > > - return err; > + return (ret == 0) ? data : ret; > } > > /* > - * Hardware-specific device functions, to implement special > - * IOCTLs for the SoundScape card. This is how we upload > - * the microcode into the card, for example, and so we > - * must ensure that no two processes can open this device > - * simultaneously, and that we can't open it at all if > - * someone is using the MIDI device. > + * Upload the microcode into the SoundScape. > */ > -static int sscape_hw_open(struct snd_hwdep * hw, struct file *file) > +static int sscape_upload_microcode(struct snd_card *card, int version) > { > - register struct soundscape *sscape = get_hwdep_soundscape(hw); > - unsigned long flags; > + struct soundscape *sscape = get_card_soundscape(card); > + const struct firmware *init_fw = NULL; > + char name[14] = "sndscape.co0"; > int err; > > - spin_lock_irqsave(&sscape->fwlock, flags); > + name[11] = '0' + version; > > - if ((sscape->midi_usage != 0) || sscape->hw_in_use) { > - err = -EBUSY; > - } else { > - sscape->hw_in_use = 1; > - err = 0; > + err = request_firmware(&init_fw, name, card->dev); > + if (err < 0) { > + snd_printk(KERN_ERR "Error loading sndscape.co0"); > + return err; > } > + err = upload_dma_data(sscape, init_fw->data, init_fw->size); > + if (err == 0) > + snd_printk(KERN_INFO "MIDI firmware loaded %d KBsn", > + init_fw->size >> 10); > > - spin_unlock_irqrestore(&sscape->fwlock, flags); > - return err; > -} > - > -static int sscape_hw_release(struct snd_hwdep * hw, struct file *file) > -{ > - register struct soundscape *sscape = get_hwdep_soundscape(hw); > - unsigned long flags; > - > - spin_lock_irqsave(&sscape->fwlock, flags); > - sscape->hw_in_use = 0; > - spin_unlock_irqrestore(&sscape->fwlock, flags); > - return 0; > -} > - > -static int sscape_hw_ioctl(struct snd_hwdep * hw, struct file *file, > - unsigned int cmd, unsigned long arg) > -{ > - struct soundscape *sscape = get_hwdep_soundscape(hw); > - int err = -EBUSY; > - > - switch (cmd) { > - case SND_SSCAPE_LOAD_BOOTB: > - { > - register struct sscape_bootblock __user *bb = (struct sscape_bootblock > __user *) arg; > - > - /* > - * We are going to have to copy this data into a special > - * DMA-able buffer before we can upload it. We shall therefore > - * just check that the data pointer is valid for now ... > - */ > - if ( !access_ok(VERIFY_READ, bb->code, sizeof(bb->code)) ) > - return -EFAULT; > - > - /* > - * Now check that we can write the firmware version number too... > - */ > - if ( !access_ok(VERIFY_WRITE, &bb->version, sizeof(bb->version)) ) > - return -EFAULT; > - > - err = sscape_upload_bootblock(sscape, bb); > - } > - break; > - > - case SND_SSCAPE_LOAD_MCODE: > - { > - register const struct sscape_microcode __user *mc = (const struct > sscape_microcode __user *) arg; > - > - err = sscape_upload_microcode(sscape, mc); > - } > - break; > - > - default: > - err = -EINVAL; > - break; > - } /* switch */ > + release_firmware(init_fw); > > return err; > } > > - > /* > * Mixer control for the SoundScape's MIDI device. > */ > @@ -920,7 +803,7 @@ static int mpu401_open(struct snd_mpu401 * mpu) > > spin_lock_irqsave(&sscape->fwlock, flags); > > - if (sscape->hw_in_use || (sscape->midi_usage == ULONG_MAX)) { > + if (sscape->midi_usage == ULONG_MAX) { > err = -EBUSY; > } else { > ++(sscape->midi_usage); > @@ -1053,13 +936,6 @@ static int __devinit create_ad1845(struct snd_card > *card, unsigned port, > } > } > > - strcpy(card->driver, "SoundScape"); > - strcpy(card->shortname, pcm->name); > - snprintf(card->longname, sizeof(card->longname), > - "%s at 0x%lx, IRQ %d, DMA1 %d, DMA2 %dn", > - pcm->name, chip->port, chip->irq, > - chip->dma1, chip->dma2); > - > sscape->chip = chip; > } > > @@ -1162,29 +1038,6 @@ static int __devinit create_sscape(int dev, struct > snd_card *card) > return -ENXIO; > } > > - if (sscape->type != SSCAPE_VIVO) { > - /* > - * Now create the hardware-specific device so that we can > - * load the microcode into the on-board processor. > - * We cannot use the MPU-401 MIDI system until this firmware > - * has been loaded into the card. > - */ > - err = snd_hwdep_new(card, "MC68EC000", 0, &(sscape->hw)); > - if (err < 0) { > - printk(KERN_ERR "sscape: Failed to create " > - "firmware devicen"); > - goto _release_dma; > - } > - strlcpy(sscape->hw->name, "SoundScape M68K", > - sizeof(sscape->hw->name)); > - sscape->hw->name[sizeof(sscape->hw->name) - 1] = '�'; > - sscape->hw->iface = SNDRV_HWDEP_IFACE_SSCAPE; > - sscape->hw->ops.open = sscape_hw_open; > - sscape->hw->ops.release = sscape_hw_release; > - sscape->hw->ops.ioctl = sscape_hw_ioctl; > - sscape->hw->private_data = sscape; > - } > - > /* > * Tell the on-board devices where their resources are (I think - > * I can't be sure without a datasheet ... So many magic values!) > @@ -1222,28 +1075,51 @@ static int __devinit create_sscape(int dev, struct > snd_card *card) > wss_port[dev], irq[dev]); > goto _release_dma; > } > + strcpy(card->driver, "SoundScape"); > + strcpy(card->shortname, name); > + snprintf(card->longname, sizeof(card->longname), > + "%s at 0x%lx, IRQ %d, DMA1 %d, DMA2 %dn", > + name, sscape->chip->port, sscape->chip->irq, > + sscape->chip->dma1, sscape->chip->dma2); > + > #define MIDI_DEVNUM 0 > if (sscape->type != SSCAPE_VIVO) { > - err = create_mpu401(card, MIDI_DEVNUM, port[dev], mpu_irq[dev]); > - if (err < 0) { > - printk(KERN_ERR "sscape: Failed to create " > - "MPU-401 device at 0x%lxn", > - port[dev]); > - goto _release_dma; > - } > + err = sscape_upload_bootblock(card); > + if (err >= 0) > + err = sscape_upload_microcode(card, err); > + > + if (err == 0) { > + err = create_mpu401(card, MIDI_DEVNUM, port[dev], > + mpu_irq[dev]); > + if (err < 0) { > + printk(KERN_ERR "sscape: Failed to create " > + "MPU-401 device at 0x%lxn", > + port[dev]); > + goto _release_dma; > + } > > - /* > - * Enable the master IRQ ... > - */ > - sscape_write(sscape, GA_INTENA_REG, 0x80); > - > - /* > - * Initialize mixer > - */ > - sscape->midi_vol = 0; > - host_write_ctrl_unsafe(sscape->io_base, CMD_SET_MIDI_VOL, 100); > - host_write_ctrl_unsafe(sscape->io_base, 0, 100); > - host_write_ctrl_unsafe(sscape->io_base, CMD_XXX_MIDI_VOL, 100); > + /* > + * Initialize mixer > + */ > + spin_lock_irqsave(&sscape->lock, flags); > + sscape->midi_vol = 0; > + host_write_ctrl_unsafe(sscape->io_base, > + CMD_SET_MIDI_VOL, 100); > + host_write_ctrl_unsafe(sscape->io_base, > + sscape->midi_vol, 100); > + host_write_ctrl_unsafe(sscape->io_base, > + CMD_XXX_MIDI_VOL, 100); > + host_write_ctrl_unsafe(sscape->io_base, > + sscape->midi_vol, 100); > + host_write_ctrl_unsafe(sscape->io_base, > + CMD_SET_EXTMIDI, 100); > + host_write_ctrl_unsafe(sscape->io_base, > + 0, 100); > + host_write_ctrl_unsafe(sscape->io_base, 80, 1); > + > + set_midi_mode_unsafe(sscape->io_base); > + spin_unlock_irqrestore(&sscape->lock, flags); > + } > } > > /* > @@ -1301,11 +1177,12 @@ static int __devinit snd_sscape_probe(struct > device *pdev, unsigned int dev) > sscape->type = SSCAPE; > > dma[dev] &= 0x03; > + snd_card_set_dev(card, pdev); > + > ret = create_sscape(dev, card); > if (ret < 0) > goto _release_card; > > - snd_card_set_dev(card, pdev); > if ((ret = snd_card_register(card)) < 0) { > printk(KERN_ERR "sscape: Failed to register sound cardn"); > goto _release_card; > @@ -1426,12 +1303,12 @@ static int __devinit sscape_pnp_detect(struct > pnp_card_link *pcard, > wss_port[idx] = pnp_port_start(dev, 1); > dma2[idx] = pnp_dma(dev, 1); > } > + snd_card_set_dev(card, &pcard->card->dev); > > ret = create_sscape(idx, card); > if (ret < 0) > goto _release_card; > > - snd_card_set_dev(card, &pcard->card->dev); > if ((ret = snd_card_register(card)) < 0) { > printk(KERN_ERR "sscape: Failed to register sound cardn"); > goto _release_card; > -- > 1.6.0.3 > > ---------------------------------------------------------------------- > Co godzine losujemy 15 000 z ---------------------------------------------------------------------- Praca za granic�? 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