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

[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/ _________________________________________________________

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[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

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[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.

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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.

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[alsa-devel] [PATCH] ARM: DaVinci: Audio support for DA850/OMAP-L138 EVM
by Chaithrika U S 21 Jan '10

21 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

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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.

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[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 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] Questions regarding workaround for VMware
by Bankim Bhavsar 18 Sep '09

18 Sep '09

Takashi I came across a workaround for VMware in ALSA kernel code http://git.kernel.org/?p=linux/kernel/git/torvalds/linux-2.6.git;a=commit;h… Thanks for the fix. In the comments you mention about inaccurate timer source. Could you elaborate on the problem? As of now we see that sound apps running in Fedora 11 guest OS with kernel 2.6.29.4-167 don't make any progress and can hang window manager like Metacity. VMware virtualizes Ensoniq ES1371 sound device. Is there something specific you would like us to fix in our virtual sound device or timer source? Thanks, Bankim.

4 6

[alsa-devel] ALSA 1.0.21 release
by Jaroslav Kysela 11 Sep '09

11 Sep '09

Hello all, all ALSA packages except alsa-oss and alsa-firmware (no changes from 1.0.20) are available for download at http://www.alsa-project.org . Changelog is available here: http://www.alsa-project.org/main/index.php/Changes_v1.0.20_v1.0.21 Jaroslav Kysela ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

3 5

[alsa-devel] [PATCH] [RFC 8/13] Intel SST sound card driver
by Vinod Koul 11 Sep '09

11 Sep '09

This adds support for Moorestown ALSA Sound card driver. This is an ALSA driver for supporting PCM playback/capture in traditional ALSA way. Anyone who chooses not to use DSP for decoding/encoding can use ALSA path to play/capture (in non low power mode). This driver registers the control interface and PCM interface with the LPE driver which finally sends it to the hardware. This driver allows any subsystem in OS which wants to use the audio-subsystems to be routed through the ALSA Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> new file: sound/pci/sst/intelmid.c --- sound/pci/sst/intelmid.c | 1761 ++++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 1761 insertions(+), 0 deletions(-) create mode 100644 sound/pci/sst/intelmid.c diff --git a/sound/pci/sst/intelmid.c b/sound/pci/sst/intelmid.c new file mode 100644 index 0000000..28a6dcc --- /dev/null +++ b/sound/pci/sst/intelmid.c @@ -0,0 +1,1761 @@ +/* + * intelmid.c - Intel Sound card driver for MID + * + * Copyright (C) 2008-09 Intel Corp + * Authors: Harsha Priya <priya.harsha(a)intel.com> + * Vinod Koul <vinod.koul(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * ALSA driver for Intel MID sound card chipset + */ +#include <sound/core.h> +#include <sound/control.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/info.h> +#include <sound/initval.h> +#include <sound/pcm-indirect.h> +#include <sound/intel_sst.h> +#include <sound/intel_sst_ioctl.h> +#include "intelmid_snd_control.h" +#include "intelmid.h" +#include "intelmid_pvt.h" +#include <linux/spi/spi.h> +#include <linux/io.h> +#include <linux/delay.h> +#include <linux/interrupt.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/moduleparam.h> +#include <linux/sched.h> +#include <linux/gpe.h> +#include <net/netlink.h> +#include <net/genetlink.h> + +MODULE_AUTHOR("Harsha Priya <priya.harsha(a)intel.com>"); +MODULE_AUTHOR("Vinod Koul <vinod.koul(a)intel.com>"); +MODULE_DESCRIPTION("Intel MAD Sound card driver"); +MODULE_LICENSE("GPL v2"); +MODULE_SUPPORTED_DEVICE("{Intel,Intel_MAD}"); + +static int card_index = SNDRV_DEFAULT_IDX1;/* Index 0-MAX */ +static char *card_id = SNDRV_DEFAULT_STR1; /* ID for this card */ +static int vendor_id; +static unsigned int audio_event_seqnum; +static int cum_bytes; + + +#ifdef FULL_CTRL +static char *out_names[] = {"Headphones", + "Internal speakers", + "EarPiece"}; +static char *in_names[] = { "DMIC", + "MIC1", + "MIC2", + "Line_in"}; +#endif +static struct genl_family audio_event_genl_family = { + .id = GENL_ID_GENERATE, + .name = "audio events", + .version = 0x01, + .maxattr = 0, +}; + +static struct genl_multicast_group audio_event_mcgrp = { + .name = "audio_group", +}; + +/*control path functionalities*/ +/** +* snd_intelmad_mute_info - provides information about the mute controls +* @kcontrol: pointer to the control +* @uinfo: pointer to the structure where the control's info need +* to be filled +* This function is called when a mixer application requests for control's info +*/ +static int snd_intelmad_mute_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + WARN_ON(!uinfo); + WARN_ON(!kcontrol); + + /*set up the mute as a boolean mono control with min-max values*/ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = MONO_CNTL; + uinfo->value.integer.min = MIN_MUTE; + uinfo->value.integer.max = MAX_MUTE; + return 0; +} + +#ifdef FULL_CTRL +/** +* snd_intelmad_volume_line_info - provides information about the volume control +* @kcontrol: pointer to the control +* @uinfo: pointer to the structure where the control's info need +* to be filled +* This function is called when a mixer application requests for control's info +*/ +static int snd_intelmad_line_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) + +{ + snd_intelmad_volume_info(uinfo, STEREO_CNTL, + ctrl_val[vendor_id].linein_vol_max, + ctrl_val[vendor_id].linein_vol_min); + return 0; +} + +/** +* snd_intelmad_mic1_volume_info - provides information about the volume control +* @kcontrol: pointer to the control +* @uinfo: pointer to the structure where the control's info need +* to be filled +* This function is called when a mixer application requests for control's info +*/ +static int snd_intelmad_mic1_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + snd_intelmad_volume_info(uinfo, STEREO_CNTL, + ctrl_val[vendor_id].mic1_vol_max, + ctrl_val[vendor_id].mic1_vol_min); + return 0; +} + +/** +* snd_intelmad_mic2_volume_info - provides information about the volume control +* @kcontrol: pointer to the control +* @uinfo: pointer to the structure where the control's info need +* to be filled +* This function is called when a mixer application requests for control's info +*/ +static int snd_intelmad_mic2_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + snd_intelmad_volume_info(uinfo, STEREO_CNTL, + ctrl_val[vendor_id].mic2_vol_max, + ctrl_val[vendor_id].mic2_vol_min); + return 0; +} + +/** +* snd_intelmad_dmic_volume_info - provides information about the volume control +* @kcontrol: pointer to the control +* @uinfo: pointer to the structure where the control's info need +* to be filled +* This function is called when a mixer application requests for control's info +*/ +static int snd_intelmad_dmic_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + snd_intelmad_volume_info(uinfo, STEREO_CNTL, + ctrl_val[vendor_id].dmic_vol_max, + ctrl_val[vendor_id].dmic_vol_min); + return 0; +} +#endif +/** +* snd_intelmad_hp_volume_info - provides information about the volume control +* @kcontrol: pointer to the control +* @uinfo: pointer to the structure where the control's info need +* to be filled +* This function is called when a mixer application requests for control's info +*/ +static int snd_intelmad_hp_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + snd_intelmad_volume_info(uinfo, STEREO_CNTL, + ctrl_val[vendor_id].hp_vol_max, + ctrl_val[vendor_id].hp_vol_min); + return 0; +} + +#ifdef FULL_CTRL +/** +* snd_intelmad_speaker_volume_info - provides information about the volume control +* @kcontrol: pointer to the control +* @uinfo: pointer to the structure where the control's info need +* to be filled +* This function is called when a mixer application requests for control's info +*/ +static int snd_intelmad_speaker_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + snd_intelmad_volume_info(uinfo, STEREO_CNTL, + ctrl_val[vendor_id].speaker_vol_max, + ctrl_val[vendor_id].speaker_vol_min); + return 0; +} + +/** +* snd_intelmad_earpiece_volume_info - provides information about the volume control +* @kcontrol: pointer to the control +* @uinfo: pointer to the structure where the control's info need +* to be filled +* This function is called when a mixer application requests for control's info +*/ +static int snd_intelmad_earpiece_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + snd_intelmad_volume_info(uinfo, MONO_CNTL, + ctrl_val[vendor_id].earpiece_vol_max, + ctrl_val[vendor_id].earpiece_vol_min); + return 0; +} + +/** +* snd_intelmad_device_info - provides information about the devices available +* @kcontrol: pointer to the control +* @uinfo: pointer to the structure where the devices's info need +* to be filled +* This function is called when a mixer application requests for device's info +*/ +static int snd_intelmad_device_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + WARN_ON(!kcontrol); + WARN_ON(!uinfo); + /*setup device select as drop down controls with different values*/ + if (OUTPUT_SEL == kcontrol->id.numid) + uinfo->value.enumerated.items = ARRAY_SIZE(out_names); + else + uinfo->value.enumerated.items = ARRAY_SIZE(in_names); + uinfo->count = MONO_CNTL; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) + uinfo->value.enumerated.item = 1; + if (OUTPUT_SEL == kcontrol->id.numid) + strncpy(uinfo->value.enumerated.name, + out_names[uinfo->value.enumerated.item], + strlen(out_names[uinfo->value.enumerated.item])); + else + strncpy(uinfo->value.enumerated.name, + in_names[uinfo->value.enumerated.item], + strlen(in_names[uinfo->value.enumerated.item])); + return 0; +} +#endif + +/** +* snd_intelmad_volume_get - gets the current volume for the control +* @kcontrol: pointer to the control +* @uval: pointer to the structure where the control's info need +* to be filled +* This function is called when .get function of a control is invoked from app +*/ +static int snd_intelmad_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uval) +{ + int ret_val = 0, cntl_list[2] = {0,}, value = 0; + struct snd_intelmad *intelmaddata = NULL; + struct snd_pmic_ops *scard_ops = NULL; + + sst_dbg("called\n"); + + WARN_ON(!uval); + WARN_ON(!kcontrol); + + intelmaddata = kcontrol->private_data; + + WARN_ON(!intelmaddata->sstdrv_ops); + + if (PMIC_UNINIT == intelmaddata->pmic_status) { + ret_val = intelmaddata->sstdrv_ops->scard_ops->init_card(); + intelmaddata->pmic_status = PMIC_INIT; + } + + if (0 != ret_val) + return ret_val; + + scard_ops = intelmaddata->sstdrv_ops->scard_ops; + + WARN_ON(!scard_ops); + + switch (kcontrol->id.numid) { + case HEADPHONES_VOL: + cntl_list[0] = PMIC_SND_RIGHT_HP_VOL; + cntl_list[1] = PMIC_SND_LEFT_HP_VOL; + break; +#ifdef FULL_CTRL + case INTERNAL_SPEAKERS_VOL: + cntl_list[0] = PMIC_SND_RIGHT_SPEAKER_VOL; + cntl_list[1] = PMIC_SND_LEFT_SPEAKER_VOL; + break; + case LINE_IN_VOL: + cntl_list[0] = PMIC_SND_INPUT_VOL_RIGHT_LINE_IN; + cntl_list[1] = PMIC_SND_INPUT_VOL_LEFT_LINE_IN; + break; + case EARPIECE_VOL: + cntl_list[0] = PMIC_SND_MONO_EARPIECE_VOL; + break; + case MIC1_VOL: + cntl_list[0] = PMIC_SND_INPUT_VOL_MIC1; + break; + case MIC2_VOL: + cntl_list[0] = PMIC_SND_INPUT_VOL_MIC2; + break; + case DMIC_VOL: + cntl_list[0] = PMIC_SND_INPUT_VOL_DMIC; + break; +#endif + default: + return -EINVAL; + } + + ret_val = scard_ops->get_vol(cntl_list[0], &value); + uval->value.integer.value[0] = value; + + if (0 != ret_val) + return ret_val; +#ifdef FULL_CTRL + if (HEADPHONES_VOL == kcontrol->id.numid + || INTERNAL_SPEAKERS_VOL == kcontrol->id.numid || + LINE_IN_VOL == kcontrol->id.numid) { +#else + if (1) { +#endif + + ret_val = scard_ops->get_vol(cntl_list[1], &value); + uval->value.integer.value[1] = value; + + if (0 != ret_val) + return ret_val; + } + return ret_val; +} + +/** +* snd_intelmad_mute_get - gets the current mute status for the control +* @kcontrol: pointer to the control +* @uval: pointer to the structure where the control's info need +* to be filled +* This function is called when .get function of a control is invoked from app +*/ +static int snd_intelmad_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uval) +{ + + int cntl_list = 0, ret_val = 0, value = 0; + struct snd_intelmad *intelmaddata = NULL; + struct snd_pmic_ops *scard_ops = NULL; + + sst_dbg("called\n"); + + WARN_ON(!uval); + WARN_ON(!kcontrol); + + intelmaddata = kcontrol->private_data; + + WARN_ON(!intelmaddata->sstdrv_ops); + + if (PMIC_UNINIT == intelmaddata->pmic_status) { + ret_val = intelmaddata->sstdrv_ops->scard_ops->init_card(); + intelmaddata->pmic_status = PMIC_INIT; + } + + if (0 != ret_val) + return ret_val; + + scard_ops = intelmaddata->sstdrv_ops->scard_ops; + + WARN_ON(!scard_ops); + + switch (kcontrol->id.numid) { + case HEADPHONES_MUTE: + cntl_list = PMIC_SND_LEFT_HP_MUTE; + break; +#ifdef FULL_CTRL + case INTERNAL_SPEAKERS_MUTE: + cntl_list = PMIC_SND_LEFT_SPEAKER_MUTE; + break; + case EARPIECE_MUTE: + cntl_list = PMIC_SND_MONO_EARPIECE_MUTE; + break; + case LINE_IN_MUTE: + cntl_list = PMIC_SND_INPUT_MUTE_LINE_IN; + break; + case MIC1_MUTE: + cntl_list = PMIC_SND_INPUT_MUTE_MIC1; + break; + case MIC2_MUTE: + cntl_list = PMIC_SND_INPUT_MUTE_MIC2; + break; + case DMIC_MUTE: + cntl_list = PMIC_SND_INPUT_MUTE_DMIC; + break; + case MASTER_MUTE: + uval->value.integer.value[0] = kcontrol->private_value; + return 0; +#endif + + default: + return -EINVAL; + } + + ret_val = scard_ops->get_mute(cntl_list, &value); + uval->value.integer.value[0] = value; + return ret_val; +} + +/** +* snd_intelmad_volume_set - sets the volume control's info +* @kcontrol: pointer to the control +* @uval: pointer to the structure where the control's info is +* available to be set +* This function is called when .set function of a control is invoked from app +*/ +static int snd_intelmad_volume_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uval) +{ + + int ret_val = 0, cntl_list[2] = {0,}; + struct snd_intelmad *intelmaddata = NULL; + struct snd_pmic_ops *scard_ops = NULL; + + sst_dbg("volume set called\n:%ld %ld", + uval->value.integer.value[0], + uval->value.integer.value[1]); + + WARN_ON(!uval); + WARN_ON(!kcontrol); + + intelmaddata = kcontrol->private_data; + + WARN_ON(!intelmaddata->sstdrv_ops); + + if (PMIC_UNINIT == intelmaddata->pmic_status) { + ret_val = intelmaddata->sstdrv_ops->scard_ops->init_card(); + intelmaddata->pmic_status = PMIC_INIT; + } + + if (0 != ret_val) + return ret_val; + + scard_ops = intelmaddata->sstdrv_ops->scard_ops; + + WARN_ON(!scard_ops); + + switch (kcontrol->id.numid) { + case HEADPHONES_VOL: + cntl_list[0] = PMIC_SND_RIGHT_HP_VOL; + cntl_list[1] = PMIC_SND_LEFT_HP_VOL; + break; +#ifdef FULL_CTRL + case INTERNAL_SPEAKERS_VOL: + cntl_list[0] = PMIC_SND_RIGHT_SPEAKER_VOL; + cntl_list[1] = PMIC_SND_LEFT_SPEAKER_VOL; + break; + case LINE_IN_VOL: + cntl_list[0] = PMIC_SND_INPUT_VOL_RIGHT_LINE_IN; + cntl_list[1] = PMIC_SND_INPUT_VOL_LEFT_LINE_IN; + break; + case EARPIECE_VOL: + cntl_list[0] = PMIC_SND_MONO_EARPIECE_VOL; + break; + case MIC1_VOL: + cntl_list[0] = PMIC_SND_INPUT_VOL_MIC1; + break; + case MIC2_VOL: + cntl_list[0] = PMIC_SND_INPUT_VOL_MIC2; + break; + case DMIC_VOL: + cntl_list[0] = PMIC_SND_INPUT_VOL_DMIC; + break; +#endif + default: + return -EINVAL; + } + + ret_val = scard_ops->set_vol(cntl_list[0], + uval->value.integer.value[0]); + if (0 != ret_val) + return ret_val; +#ifdef FULL_CTRL + if (HEADPHONES_VOL == kcontrol->id.numid + || INTERNAL_SPEAKERS_VOL == kcontrol->id.numid || + LINE_IN_VOL == kcontrol->id.numid) { +#else + if (1) { +#endif + ret_val = scard_ops->set_vol(cntl_list[1], + uval->value.integer.value[1]); + if (0 != ret_val) + return ret_val; + } + kcontrol->private_value = uval->value.integer.value[0]; + return ret_val; +} + + + +/** +* snd_intelmad_mute_set - sets the mute control's info +* @kcontrol: pointer to the control +* @uval: pointer to the structure where the control's info is +* available to be set +* This function is called when .set function of a control is invoked from app +*/ +static int snd_intelmad_mute_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uval) +{ + int cntl_list[2] = {0,}, ret_val = 0; + struct snd_intelmad *intelmaddata = NULL; + struct snd_pmic_ops *scard_ops = NULL; + + sst_dbg("called\n"); + + WARN_ON(!uval); + WARN_ON(!kcontrol); + + intelmaddata = kcontrol->private_data; + + WARN_ON(!intelmaddata->sstdrv_ops); + + if (PMIC_UNINIT == intelmaddata->pmic_status) { + ret_val = intelmaddata->sstdrv_ops->scard_ops->init_card(); + intelmaddata->pmic_status = PMIC_INIT; + } + + if (0 != ret_val) + return ret_val; + + scard_ops = intelmaddata->sstdrv_ops->scard_ops; + + WARN_ON(!scard_ops); + + kcontrol->private_value = uval->value.integer.value[0]; + + switch (kcontrol->id.numid) { + case HEADPHONES_MUTE: + cntl_list[0] = PMIC_SND_LEFT_HP_MUTE; + cntl_list[1] = PMIC_SND_RIGHT_HP_MUTE; + break; +#ifdef FULL_CTRL + case INTERNAL_SPEAKERS_MUTE: + cntl_list[0] = PMIC_SND_LEFT_SPEAKER_MUTE; + cntl_list[1] = PMIC_SND_RIGHT_SPEAKER_MUTE; + break; + case EARPIECE_MUTE: + cntl_list[0] = PMIC_SND_MONO_EARPIECE_MUTE; + break; + case LINE_IN_MUTE: + cntl_list[0] = PMIC_SND_INPUT_MUTE_LINE_IN; + break; + case MIC1_MUTE: + cntl_list[0] = PMIC_SND_INPUT_MUTE_MIC1; + break; + case MIC2_MUTE: + cntl_list[0] = PMIC_SND_INPUT_MUTE_MIC2; + break; + case DMIC_MUTE: + cntl_list[0] = PMIC_SND_INPUT_MUTE_DMIC; + break; + case MASTER_MUTE: + mute_all(kcontrol, uval); + kcontrol->private_value = uval->value.integer.value[0]; + return 0; +#endif + default: + return -EINVAL; + } + + ret_val = scard_ops->set_mute(cntl_list[0], + uval->value.integer.value[0]); + if (0 != ret_val) + return ret_val; + +#ifdef FULL_CTRL + if (HEADPHONES_MUTE == kcontrol->id.numid + || INTERNAL_SPEAKERS_MUTE == kcontrol->id.numid) +#endif + ret_val = scard_ops->set_mute(cntl_list[1], + uval->value.integer.value[0]); + kcontrol->private_value = uval->value.integer.value[0]; + return ret_val; +} + +#ifdef FULL_CTRL +/** +* snd_intelmad_device_get - get the device select control's info +* @kcontrol: pointer to the control +* @uval: pointer to the structure where the control's info is +* to be filled +* This function is called when .get function of a control is invoked from app +*/ +static int snd_intelmad_device_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uval) +{ + sst_dbg("called\n"); + + WARN_ON(!uval); + WARN_ON(!kcontrol); + + uval->value.enumerated.item[0] = kcontrol->private_value; + return 0; +} + +/** +* snd_intelmad_device_set - set the device select control's info +* @kcontrol: pointer to the control +* @uval: pointer to the structure where the control's info is +* available to be set +* This function is called when .set function of a control is invoked from app +*/ +static int snd_intelmad_device_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uval) +{ + struct snd_intelmad *intelmaddata = NULL; + struct snd_pmic_ops *scard_ops = NULL; + int ret_val = 0; + + sst_dbg("called\n"); + + WARN_ON(!uval); + WARN_ON(!kcontrol); + + intelmaddata = kcontrol->private_data; + + WARN_ON(!intelmaddata->sstdrv_ops); + + if (PMIC_UNINIT == intelmaddata->pmic_status) { + ret_val = intelmaddata->sstdrv_ops->scard_ops->init_card(); + intelmaddata->pmic_status = PMIC_INIT; + } + + if (0 != ret_val) + return ret_val; + + scard_ops = intelmaddata->sstdrv_ops->scard_ops; + + WARN_ON(!scard_ops); + + /*store value with driver*/ + kcontrol->private_value = uval->value.enumerated.item[0]; + + switch (kcontrol->id.numid) { + case OUTPUT_SEL: + ret_val = scard_ops->set_output_dev( + uval->value.enumerated.item[0]); + break; + case INPUT_SEL: + ret_val = scard_ops->set_input_dev( + uval->value.enumerated.item[0]); + break; + default: + return -EINVAL; + } + kcontrol->private_value = uval->value.enumerated.item[0]; + return ret_val; +} +#endif + +/*structure of all the controls of the sound card that is exposed*/ +static struct snd_kcontrol_new snd_intelmad_controls[] __devinitdata = { +#ifdef FULL_CTRL +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_line_volume_info, + .get = snd_intelmad_volume_get, + .put = snd_intelmad_volume_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Capture Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_mute_info, + .get = snd_intelmad_mute_get, + .put = snd_intelmad_mute_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "DMIC Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_dmic_volume_info, + .get = snd_intelmad_volume_get, + .put = snd_intelmad_volume_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "DMIC Capture Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_mute_info, + .get = snd_intelmad_mute_get, + .put = snd_intelmad_mute_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic1 Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_mic1_volume_info, + .get = snd_intelmad_volume_get, + .put = snd_intelmad_volume_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic1 Capture Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_mute_info, + .get = snd_intelmad_mute_get, + .put = snd_intelmad_mute_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic2 Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_mic2_volume_info, + .get = snd_intelmad_volume_get, + .put = snd_intelmad_volume_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic2 Capture Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_mute_info, + .get = snd_intelmad_mute_get, + .put = snd_intelmad_mute_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Capture Source", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_device_info, + .get = snd_intelmad_device_get, + .put = snd_intelmad_device_set, + .private_value = 0, +}, +#endif +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_hp_volume_info, + .get = snd_intelmad_volume_get, + .put = snd_intelmad_volume_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone Playback Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_mute_info, + .get = snd_intelmad_mute_get, + .put = snd_intelmad_mute_set, + .private_value = 0, +}, +#ifdef FULL_CTRL +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Internal Speaker Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_speaker_volume_info, + .get = snd_intelmad_volume_get, + .put = snd_intelmad_volume_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Internal Speaker Playback Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_mute_info, + .get = snd_intelmad_mute_get, + .put = snd_intelmad_mute_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mono Earpiece Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_earpiece_volume_info, + .get = snd_intelmad_volume_get, + .put = snd_intelmad_volume_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mono Earpiece Playback Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_mute_info, + .get = snd_intelmad_mute_get, + .put = snd_intelmad_mute_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Source", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_device_info, + .get = snd_intelmad_device_get, + .put = snd_intelmad_device_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Mute Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_mute_info, + .get = snd_intelmad_mute_get, + .put = snd_intelmad_mute_set, + .private_value = 0, +}, +#endif +}; + +/*Data path functionalities*/ +static struct snd_pcm_hardware snd_intelmad_stream = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_DOUBLE | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_MMAP| + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_SYNC_START), + .formats = (SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_U16 | + /*SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 |*/ + SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_U24), + .rates = (SNDRV_PCM_RATE_8000| + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000), + .rate_min = MIN_RATE, + + .rate_max = MAX_RATE, + .channels_min = MIN_CHANNEL, + .channels_max = MAX_CHANNEL, + .buffer_bytes_max = MAX_BUFFER, + .period_bytes_min = MIN_PERIOD_BYTES, + .period_bytes_max = MAX_PERIOD_BYTES, + .periods_min = MIN_PERIODS, + .periods_max = MAX_PERIODS, + .fifo_size = FIFO_SIZE, +}; + +static void send_buffer_to_sst(struct snd_pcm_substream *substream, + struct snd_pcm_indirect *rec, size_t bytes) +{ + struct snd_intelmad *intelmaddata = snd_pcm_substream_chip(substream); + struct stream_buffer buffer_to_sst = {0,}; + struct mad_stream_pvt *stream = substream->runtime->private_data; + int ret_val = 0; + /*sends data to SST to be processed*/ + + cum_bytes += bytes; + sst_dbg("bytes = %d \n", bytes); + sst_dbg("cum_bytes = 0x%x, \n", cum_bytes); + buffer_to_sst.length = bytes; + buffer_to_sst.addr = (unsigned long) substream->runtime->dma_area + + rec->sw_data; + /*SST API to actually send the buffer to be played*/ + ret_val = intelmaddata->sstdrv_ops->send_buffer(stream->stream_id, + &buffer_to_sst); + sst_dbg("send_buffer ret_val = 0x%x \n", ret_val); + return; +} + +void start_capture(struct work_struct *work) +{ + int ret_val = 0; + struct mad_capture_wq *stream_work = + container_of(work, struct mad_capture_wq, wq); + struct stream_buffer *buffer_to_sst = &stream_work->buffer_to_sst; + struct snd_pcm_substream *substream = stream_work->substream; + struct mad_stream_pvt *stream = substream->runtime->private_data; + struct snd_intelmad *intelmaddata = snd_pcm_substream_chip(substream); + + ret_val = intelmaddata->sstdrv_ops->send_buffer(stream->stream_id, + buffer_to_sst); + sst_dbg("send_buffer ret_val = 0x%x \n", ret_val); + return; +} + +void snd_card_pcm_wq_function(struct work_struct *work) +{ + struct mad_wq *stream_work = + container_of(work, struct mad_wq, wq); + struct snd_pcm_substream *substream; + struct snd_pcm_indirect *rec = NULL; + struct mad_stream_pvt *stream; + + mutex_lock(&stream_work->wq_lock); + substream = stream_work->substream; + if (substream == NULL || substream->runtime == NULL) { + sst_err("subsream structure is null \n"); + mutex_unlock(&stream_work->wq_lock); + return; + } + stream = substream->runtime->private_data; + sst_dbg("called %d\n", stream->stream_status); + if (stream == NULL) { + sst_err("stream is null in wq function \n"); + mutex_unlock(&stream_work->wq_lock); + return; + } + + if (stream->stream_status != INIT) { + rec = &stream->pcm_indirect; + if (substream->stream == STREAM_OPS_PLAYBACK) + snd_pcm_indirect_playback_transfer(substream, rec, + send_buffer_to_sst); + else if (substream->stream == STREAM_OPS_CAPTURE) { + sst_dbg("calling indirect capture transfer\n"); + snd_pcm_indirect_capture_transfer(substream, rec, + send_buffer_to_sst); + } + stream->stream_status = RUNNING; + } + mutex_unlock(&stream_work->wq_lock); + return; +} + +static int snd_intelmad_pcm_ack(struct snd_pcm_substream *substream) +{ + struct mad_stream_pvt *stream; + + stream = substream->runtime->private_data; + sst_dbg("...called\n"); + stream->mad_msg_wq.substream = substream; + schedule_work(&stream->mad_msg_wq.wq); + return 0; +} + +/** +* snd_intelmad_pcm_trigger - stream activities are handled here +* @substream:substream for which the stream function is called +*@cmd:the stream commamd thats requested from upper layer +* This function is called whenever an a stream activity is invoked +*/ +static int snd_intelmad_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + int ret_val = 0, stream_type = 0/*, sample_size = 0*/; + struct snd_intelmad *intelmaddata = NULL; + struct mad_stream_pvt *stream = NULL; + + sst_dbg("called\n"); + + WARN_ON(!substream); + + intelmaddata = snd_pcm_substream_chip(substream); + stream_type = substream->stream; + stream = substream->runtime->private_data; + + if (PMIC_UNINIT == intelmaddata->pmic_status) { + ret_val = intelmaddata->sstdrv_ops->scard_ops->init_card(); + intelmaddata->pmic_status = PMIC_INIT; + } + + if (0 != ret_val) + return ret_val; + + WARN_ON(!intelmaddata->sstdrv_ops); + WARN_ON(!intelmaddata->sstdrv_ops->scard_ops); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + /*invoked on a power managment -resume operation*/ + /*stream parameters and context resuming to be done here*/ + /*TBD*/ + sst_dbg("in resume\n"); + break; + case SNDRV_PCM_TRIGGER_START: + stream->substream = substream; + stream->pcm_size = snd_pcm_lib_buffer_bytes(substream); + sst_dbg("pcm_size=%d\n", stream->pcm_size); +#ifdef TIMER_TS + bps = (substream->runtime->sample_bits/8) + * (substream->runtime->channels) + * (substream->runtime->rate); + stream->pcm_jiffie = ((bps * ELAPSED_PERIOD) / HZ); + stream->pcm_buf_pos = 0; + sst_dbg("pcm_jiffie=%d \n", stream->pcm_jiffie); +#endif + stream->stream_status = STARTED; + if (substream->stream == STREAM_OPS_PLAYBACK) + snd_intelmad_pcm_ack(substream); + else if (substream->stream == STREAM_OPS_CAPTURE) { + stream->mad_capture_wq.substream = substream; + stream->mad_capture_wq.buffer_to_sst.length = + frames_to_bytes(substream->runtime, + substream->runtime->period_size); + stream->mad_capture_wq.buffer_to_sst.addr = + (unsigned long) substream->runtime->dma_area; + schedule_work(&stream->mad_capture_wq.wq); + cum_bytes = stream->mad_capture_wq.buffer_to_sst.length; + stream->stream_status = RUNNING; + } + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + /*invoked on suspend - power management event*/ + /*stream parameters and context to be saved*/ + /*TBD*/ + sst_dbg("in suspend\n"); + break; + case SNDRV_PCM_TRIGGER_STOP: + sst_dbg("in stop\n"); + /*ret_val = intelmaddata->sstdrv_ops->control_set(SST_SND_DROP, + &stream->stream_id); + if (0 != ret_val) + return ret_val;*/ + stream->stream_status = DROPPED; + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + sst_dbg("in pause\n"); + ret_val = intelmaddata->sstdrv_ops->control_set(SST_SND_PAUSE, + &stream->stream_id); + if (0 != ret_val) + return ret_val; + stream->stream_status = PAUSED; + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + sst_dbg("in pause release \n"); + ret_val = intelmaddata->sstdrv_ops->control_set(SST_SND_RESUME, + &stream->stream_id); + if (0 != ret_val) + return ret_val; + stream->stream_status = RUNNING; + break; + default: + return -EINVAL; + } + return ret_val; +} + +/** +* snd_intelmad_pcm_prepare- internal preparation before starting a stream +* @substream: substream for which the function is called +* This function is called when a stream is started for internal preparation. +*/ +static int snd_intelmad_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct mad_stream_pvt *stream = NULL; + int ret_val = 0; + + sst_dbg("called \n"); + + WARN_ON(!substream); + + stream = substream->runtime->private_data; + + sst_dbg("format = %d \n", substream->runtime->format); + if (stream->stream_id == 0) { + ret_val = snd_intelmad_alloc_stream(substream); + cum_bytes = 0; + if (ret_val < 0) + return ret_val; + ret_val = snd_intelmad_init_stream(substream); + if (0 != ret_val) + return ret_val; + sst_dbg("period size = %d \n", + (int)substream->runtime->period_size); + sst_dbg("buf size = %d \n", + (int)substream->runtime->buffer_size); + memset(&stream->pcm_indirect, 0, sizeof(stream->pcm_indirect)); + stream->pcm_indirect.hw_buffer_size = + snd_pcm_lib_buffer_bytes(substream); + stream->pcm_indirect.sw_buffer_size = + stream->pcm_indirect.hw_buffer_size; + /*return back the stream id*/ + snprintf(substream->pcm->id, sizeof(substream->pcm->id), + "%d", stream->stream_id); + sst_dbg("stream id to user = %s\n", substream->pcm->id); + } + return ret_val; +} + +static int snd_intelmad_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + int ret_val = 0; + + sst_dbg("called\n"); + ret_val = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + memset(substream->runtime->dma_area, 0, params_buffer_bytes(hw_params)); + return ret_val; +} + +static int snd_intelmad_hw_free(struct snd_pcm_substream *substream) +{ + sst_dbg("called\n"); + return snd_pcm_lib_free_pages(substream); +} + +/** +* snd_intelmad_pcm_pointer- to send the current buffer pointer processed by hw +* @substream: substream for which the function is called +* This function is called by ALSA framework to get the current hw buffer ptr +* when a period is elapsed +*/ +static snd_pcm_uframes_t snd_intelmad_pcm_pointer + (struct snd_pcm_substream *substream) +{ + /*struct snd_pcm_runtime *runtime = substream->runtime;*/ + struct mad_stream_pvt *stream = NULL; + struct snd_intelmad *intelmaddata = NULL; + int ret_val = 0; + unsigned long buf_size = 0; + + WARN_ON(!substream); + + intelmaddata = snd_pcm_substream_chip(substream); + stream = substream->runtime->private_data; + if (stream->stream_status == INIT) + return 0; + + stream->buf_ptr.str_id = stream->stream_id; + +#ifndef TIMER_TS + ret_val = intelmaddata->sstdrv_ops->control_set(SST_SND_BUFFER_POINTER, + &stream->buf_ptr); + if (0 != ret_val) { + sst_err("error code = 0x%x \n", ret_val); + return ret_val; + } + sst_dbg("samples reported out 0x%lx \n", stream->buf_ptr.buffer_ptr); + buf_size = frames_to_bytes(substream->runtime, + stream->buf_ptr.buffer_ptr); + sst_dbg(" bytes reported out = 0x%lx\n", buf_size); + if (buf_size > cum_bytes) + sst_err("excess reported\n"); + + if (substream->stream == STREAM_OPS_PLAYBACK) + stream->period_elapsed_ptr = snd_pcm_indirect_playback_pointer( + substream, &stream->pcm_indirect, buf_size); + else + stream->period_elapsed_ptr = snd_pcm_indirect_capture_pointer( + substream, &stream->pcm_indirect, buf_size); + + sst_dbg("indirect value = 0x%lx\n", stream->period_elapsed_ptr); + + return stream->period_elapsed_ptr; +#else + stream->pcm_buf_pos += stream->pcm_jiffie; + stream->pcm_buf_pos %= stream->pcm_size; + return snd_pcm_indirect_playback_pointer(substream, + &stream->pcm_indirect, stream->pcm_buf_pos); +#endif + +} + +/** +* snd_intelmad_close- to free parameteres when stream is stopped +* @substream: substream for which the function is called +* This function is called by ALSA framework when stream is stopped +*/ +static int snd_intelmad_close(struct snd_pcm_substream *substream) +{ + struct snd_intelmad *intelmaddata = NULL; + struct mad_stream_pvt *stream = NULL; + int ret_val = 0; + + WARN_ON(!substream); + + stream = substream->runtime->private_data; + + sst_dbg("called \n"); + intelmaddata = snd_pcm_substream_chip(substream); + mutex_lock(&stream->mad_msg_wq.wq_lock); + sst_dbg("str id = %d\n", stream->stream_id); + /*SST API to actually stop/free the stream*/ + ret_val = intelmaddata->sstdrv_ops->control_set(SST_SND_FREE, + &stream->stream_id); + kfree(substream->runtime->private_data); + mutex_unlock(&stream->mad_msg_wq.wq_lock); + return ret_val; +} + +/** +* snd_intelmad_open- to set runtime parameters during stream start +* @substream: substream for which the function is called +* This function is called by ALSA framework when stream is started +*/ +static int snd_intelmad_open(struct snd_pcm_substream *substream) +{ + struct snd_intelmad *intelmaddata; + struct snd_pcm_runtime *runtime; + struct mad_stream_pvt *stream; + + WARN_ON(!substream); + + sst_dbg("called \n"); + + intelmaddata = snd_pcm_substream_chip(substream); + runtime = substream->runtime; + /*set the runtime hw parameter with our local snd_pcm_hardware struct*/ + runtime->hw = snd_intelmad_stream; + /*setup the internal datastruture stream pointers based on it being + playback or capture stream*/ + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (NULL == stream) + return -ENOMEM; + stream->stream_id = 0; + stream->stream_status = INIT; + INIT_WORK(&stream->mad_msg_wq.wq, snd_card_pcm_wq_function); + INIT_WORK(&stream->mad_capture_wq.wq, start_capture); + mutex_init(&stream->mad_msg_wq.wq_lock); + runtime->private_data = stream; + return 0; +} + +static struct snd_pcm_ops snd_intelmad_playback_ops = { + .open = snd_intelmad_open, + .close = snd_intelmad_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_intelmad_hw_params, + .hw_free = snd_intelmad_hw_free, + .prepare = snd_intelmad_pcm_prepare, + .trigger = snd_intelmad_pcm_trigger, + .pointer = snd_intelmad_pcm_pointer, + .ack = snd_intelmad_pcm_ack, +}; + +static struct snd_pcm_ops snd_intelmad_capture_ops = { + .open = snd_intelmad_open, + .close = snd_intelmad_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_intelmad_hw_params, + .hw_free = snd_intelmad_hw_free, + .prepare = snd_intelmad_pcm_prepare, + .trigger = snd_intelmad_pcm_trigger, + .pointer = snd_intelmad_pcm_pointer, + .ack = snd_intelmad_pcm_ack, +}; + +int snd_intelmad_generate_netlink(u32 orig, enum eaudio_events event) +{ + struct sk_buff *skb = NULL; + struct nlattr *attr = NULL; + struct audio_genl_event *aud_event = NULL; + void *msg_header = NULL; + int size = 0, ret_val = 0; + + /* allocate memory */ + size = nla_total_size(sizeof(struct audio_genl_event)) + \ + nla_total_size(0); + + skb = genlmsg_new(size, GFP_ATOMIC); + if (!skb) + return -ENOMEM; + + /* add the genetlink message header */ + msg_header = genlmsg_put(skb, 0, audio_event_seqnum++, + &audio_event_genl_family, 0, + AUDIO_GENL_CMD_EVENT); + if (!msg_header) { + nlmsg_free(skb); + return -ENOMEM; + } + + /* fill the data */ + attr = nla_reserve(skb, AUDIO_GENL_ATTR_EVENT, \ + sizeof(struct audio_genl_event)); + + if (!attr) { + nlmsg_free(skb); + return -EINVAL; + } + + aud_event = nla_data(attr); + if (!aud_event) { + nlmsg_free(skb); + return -EINVAL; + } + + memset(aud_event, 0, sizeof(struct audio_genl_event)); + + aud_event->orig = orig; + aud_event->event = event; + + /* send multicast genetlink message */ + ret_val = genlmsg_end(skb, msg_header); + if (ret_val < 0) { + nlmsg_free(skb); + return ret_val; + } + + ret_val = genlmsg_multicast(skb, 0, audio_event_mcgrp.id, GFP_ATOMIC); + + if (ret_val) + printk(KERN_INFO "Failed to send a Genetlink message!\n"); + return 0; +} + +#ifdef REG_IRQ +/** +* snd_intelmad_intr_handler- interrupt handler +*@irq : irq number of the interrupt received +*@dev: device context +* This function is called when an interrupt is raised at the sound card +*/ +static irqreturn_t snd_intelmad_intr_handler(int irq, void *dev) +{ + struct snd_intelmad *intelmaddata = + (struct snd_intelmad *)dev; + struct sc_reg_access pmic_reg = {0,}; + int ret_val = 0; + u8 intsts = 0; + memcpy_fromio(&intsts, + ((void *)(intelmaddata->int_base)), + sizeof(u8)); + if (intsts != 0) + sst_dbg("intstatus = 0x%x\n", intsts); + switch (intelmaddata->sstdrv_ops->vendor_id) { + case SND_FS: + if (intsts & 0x1) { + /*send long push*/ + ret_val = snd_intelmad_generate_netlink(1, + AUDIO_EVENT_LONG_PRESS); + } + if (intsts & 0x2) { + /*send short push*/ + ret_val = snd_intelmad_generate_netlink(1, + AUDIO_EVENT_SHORT_PRESS); + } + if (intsts & 0x4) { + /*send headphone detect*/ + ret_val = snd_intelmad_generate_netlink(1, + AUDIO_EVENT_HP_DETECT); + } + if (intsts & 0x8) { + /*send headset detect*/ + ret_val = snd_intelmad_generate_netlink(1, + AUDIO_EVENT_HS_DETECT); + } + break; + case SND_MX: + if (intsts & 0x2) { + pmic_reg.reg_addr = 0x201; + ret_val = sst_sc_reg_access(&pmic_reg, PMIC_READ, 1); + if (0 != ret_val) { + /*pmic communication fails*/ + sst_err("pmic commn failed \n"); + return IRQ_HANDLED; + } + intsts = pmic_reg.value; + if (intsts & 0xc0) + /*send jack detect/undetect*/ + sst_dbg("to handle jack sensing\n"); + } + break; + case SND_NC: + if (intsts & 0x1) { + /*send headset detect/undetect*/ + ret_val = snd_intelmad_generate_netlink(1, + AUDIO_EVENT_HS_DETECT); + } + if (intsts & 0x2) { + /*send headphone detect/undetect*/ + ret_val = snd_intelmad_generate_netlink(1, + AUDIO_EVENT_HP_DETECT); + } + if (intsts & 0x4) { + /*send short push*/ + ret_val = snd_intelmad_generate_netlink(1, + AUDIO_EVENT_SHORT_PRESS); + } + if (intsts & 0x8) { + /*send long push*/ + ret_val = snd_intelmad_generate_netlink(1, + AUDIO_EVENT_LONG_PRESS); + } + break; + } + return IRQ_HANDLED; +} +static int __devinit snd_intelmad_register_irq( + struct snd_intelmad *intelmaddata) +{ + int ret_val = 0; + u32 regbase = AUDINT_BASE, regsize = 8; + + /* interpret irq field */ + sst_dbg("irq = 0x%x\n", intelmaddata->irq); + + if ((intelmaddata->irq) & PMIC_SOUND_IRQ_TYPE_MASK) { + /* irq -> GPE_ID */ + intelmaddata->irq = intelmaddata->irq & + (~PMIC_SOUND_IRQ_TYPE_MASK); + sst_dbg("gpe = 0x%x\n", intelmaddata->irq); + ret_val = request_gpe(intelmaddata->irq, + (gpio_function_t)snd_intelmad_intr_handler, + intelmaddata, 0); + if (ret_val) { + sst_dbg("cannot register gpe\n"); + return ret_val; + } + + } else { + /* irq -> system irq */ + ret_val = request_irq(intelmaddata->irq, + snd_intelmad_intr_handler, + IRQF_SHARED, DRIVER_NAME, + intelmaddata); + if (0 != ret_val) { + sst_dbg("cannot register IRQ\n"); + return ret_val; + } + } + + intelmaddata->int_base = ioremap_nocache(regbase, regsize); + return ret_val; +} +static int __devinit snd_intelmad_register_netlink(void) +{ + int ret_val = 0; + + ret_val = genl_register_family(&audio_event_genl_family); + if (0 != ret_val) { + sst_dbg("netlink registration failed\n"); + return ret_val; + } + ret_val = genl_register_mc_group(&audio_event_genl_family, + &audio_event_mcgrp); + if (ret_val) { + sst_dbg("netlink group registration failed\n"); + genl_unregister_family(&audio_event_genl_family); + return ret_val; + } + return ret_val; +} +#endif + +static int __devinit snd_intelmad_sst_register( + struct snd_intelmad *intelmaddata) +{ + int ret_val = 0; + struct sc_reg_access pmic_reg = {0,}; + + pmic_reg.reg_addr = 0; + ret_val = sst_sc_reg_access(&pmic_reg, PMIC_READ, 1); + + if (0 != ret_val) + return ret_val; + + vendor_id = pmic_reg.value & (MASK2|MASK1|MASK0); + sst_info("orginal reg n extrated vendor id = 0x%x %d\n", + pmic_reg.value, vendor_id); + if (vendor_id < 0 || vendor_id > 2) { + sst_err("vendor card not supported!!\n"); + return -EIO; + } + intelmaddata->sstdrv_ops->module_name = SST_CARD_NAMES; + intelmaddata->sstdrv_ops->vendor_id = vendor_id; + intelmaddata->sstdrv_ops->scard_ops = &v[vendor_id]; + + /*registering with SST driver to get access to SST APIs to use*/ + ret_val = register_sst_card(intelmaddata->sstdrv_ops); + if (0 != ret_val) { + sst_err("sst card registration failed\n"); + return ret_val; + } + + vendor_id = intelmaddata->sstdrv_ops->vendor_id; + intelmaddata->pmic_status = PMIC_UNINIT; + return ret_val; +} + +/*Driver Init/exit functionalities*/ +/** +* snd_intelmad_pcm- to setup pcm for the card +* @card: pointer to the sound card structure +*@intelmaddata: pointer to internal context +* This function is called from probe function to set up pcm params and functions +*/ +static int __devinit snd_intelmad_pcm(struct snd_card *card, + struct snd_intelmad *intelmaddata) +{ + struct snd_pcm *pcm; + int i, ret_val = 0; + char name[32] = INTEL_MAD; + + WARN_ON(!card); + WARN_ON(!intelmaddata); + + for (i = 0; i < MAX_DEVICES; i++) { + ret_val = snd_pcm_new(card, name, i, PLAYBACK_COUNT, + CAPTURE_COUNT, &pcm); + if (0 != ret_val) + break; + /*setup the ops for playback and capture streams*/ + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_intelmad_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_intelmad_capture_ops); + /*setup private data which can be retrieved when required*/ + pcm->private_data = intelmaddata; + pcm->info_flags = 0; + strncpy(pcm->name, card->shortname, strlen(card->shortname)); + /*allocate dma pages for ALSA stream operations*/ + snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + MIN_BUFFER, MAX_BUFFER); + } + return ret_val; +} + +/** +* snd_intelmad_mixer- to setup mixer settings of the card +*@intelmaddata: pointer to internal context +* This function is called from probe function to set up mixer controls +*/ +static int __devinit snd_intelmad_mixer(struct snd_intelmad *intelmaddata) +{ + struct snd_card *card = NULL; + unsigned int idx; + int ret_val = 0; + char *mixername = "IntelMAD Controls"; + + WARN_ON(!intelmaddata); + + card = intelmaddata->card; + + strncpy(card->mixername, mixername, strlen(mixername)); + /*add all widget controls and expose the same*/ + for (idx = 0; idx < ARRAY_SIZE(snd_intelmad_controls); idx++) { + ret_val = snd_ctl_add(card, + snd_ctl_new1(&snd_intelmad_controls[idx], + intelmaddata)); + sst_dbg("mixer[idx]=%d added \n", idx); + if (0 != ret_val) + break; + } + return ret_val; +} + +/** +* snd_intelmad_dev_free- to free device +*@device: pointer to the device +* This function is called when driver module is removed +*/ +static int snd_intelmad_dev_free(struct snd_device *device) +{ + struct snd_intelmad *intelmaddata; + + WARN_ON(!device); + + intelmaddata = device->device_data; + + sst_dbg("called\n"); + snd_card_free(intelmaddata->card); + genl_unregister_family(&audio_event_genl_family); + unregister_sst_card(intelmaddata->sstdrv_ops); + + /*free allocated memory for internal context*/ + kfree(intelmaddata->sstdrv_ops); + kfree(intelmaddata); + return 0; +} + +/** +* snd_intelmad_create- called from probe to create a snd device +*@intelmaddata : pointer to the internal context +*@card : pointer to the sound card +* This function is called when driver module is started +*/ +static int __devinit snd_intelmad_create( + struct snd_intelmad *intelmaddata, + struct snd_card *card) +{ + int ret_val = 0; + static struct snd_device_ops ops = { + .dev_free = snd_intelmad_dev_free, + }; + + WARN_ON(!intelmaddata); + WARN_ON(!card); + /*ALSA api to register for the device*/ + ret_val = snd_device_new(card, SNDRV_DEV_LOWLEVEL, intelmaddata, &ops); + return ret_val; +} + +/********************************************************************* + * SPI Functions + *********************************************************************/ + + +/** +* snd_intelmad_probe- function registred for init +*@spi : pointer to the spi device context +* This function is called when the device is initialized +*/ +int __devinit snd_intelmad_probe(struct spi_device *spi) +{ + struct snd_card *card = NULL; + int ret_val = 0; + struct snd_intelmad *intelmaddata = NULL; + + sst_dbg("called \n"); + + /*allocate memory for saving internal context and working*/ + intelmaddata = kzalloc(sizeof(*intelmaddata), GFP_KERNEL); + if (NULL == intelmaddata) + return -ENOMEM; + + /*allocate memory for LPE API set*/ + intelmaddata->sstdrv_ops = kzalloc(sizeof(struct intel_sst_card_ops), + GFP_KERNEL); + if (NULL == intelmaddata->sstdrv_ops) { + sst_err("mem alloctn fail \n"); + kfree(intelmaddata); + return -ENOMEM; + } + + /*create a card instance with ALSA framework*/ + card = snd_card_new(card_index, card_id, THIS_MODULE, 0); + if (NULL == card) + return -ENOMEM; + + intelmaddata->spi = spi; + intelmaddata->irq = spi->irq; + dev_set_drvdata(&spi->dev, intelmaddata); + intelmaddata->card = card; + intelmaddata->card_id = card_id; + intelmaddata->card_index = card_index; + strncpy(card->driver, INTEL_MAD, strlen(INTEL_MAD)); + strncpy(card->shortname, INTEL_MAD, strlen(INTEL_MAD)); + +#ifdef REG_IRQ + ret_val = snd_intelmad_register_irq(intelmaddata); + if (0 != ret_val) { + sst_err("snd_intelmad_register_irq fail\n"); + goto free_allocs; + } + ret_val = snd_intelmad_register_netlink(); + if (0 == ret_val) { + sst_dbg("...complete\n"); + return ret_val; + } +#endif + + /*internal function call to register device with ALSA*/ + ret_val = snd_intelmad_create(intelmaddata, card); + if (0 != ret_val) { + sst_err("snd_intelmad_create failed\n"); + goto free_allocs; + } + + ret_val = snd_intelmad_pcm(card, intelmaddata); + if (0 != ret_val) { + sst_err("snd_intelmad_pcm failed\n"); + goto free_allocs; + } + + ret_val = snd_intelmad_mixer(intelmaddata); + if (0 != ret_val) { + sst_err("snd_intelmad_mixer failed\n"); + goto free_allocs; + } + + card->private_data = &intelmaddata; + snd_card_set_dev(card, &spi->dev); + ret_val = snd_card_register(card); + if (0 != ret_val) + goto free_allocs; + + + intelmaddata->sstdrv_ops->module_name = SST_CARD_NAMES; + /*registering with LPE driver to get access to LPE APIsito use*/ + ret_val = snd_intelmad_sst_register(intelmaddata); + if (0 == ret_val) { + ret_val = intelmaddata->sstdrv_ops->scard_ops->init_card(); + intelmaddata->pmic_status = PMIC_INIT; + sst_dbg("...complete\n"); + return ret_val; + } + + sst_err("registering with LPE failed \n"); + +free_allocs: + /*TODO: unregister IRQ*/ + sst_err("probe failed\n"); + /*snd_card_free(card);*/ + kfree(intelmaddata->sstdrv_ops); + kfree(intelmaddata); + return ret_val; +} + + +/** +* snd_intelmad_remove- function registred for exit +*@spi : pointer to the spi device context +* This function is called when the device is uninitialized +*/ +static int snd_intelmad_remove(struct spi_device *spi) +{ + struct snd_intelmad *intelmaddata = + dev_get_drvdata(&spi->dev); + /* + * TODO:: de-register interrupt handler + */ + + if (intelmaddata) { + snd_card_free(intelmaddata->card); + genl_unregister_family(&audio_event_genl_family); + unregister_sst_card(intelmaddata->sstdrv_ops); + /*free allocated memory for internal context*/ + kfree(intelmaddata->sstdrv_ops); + kfree(intelmaddata); + } + return 0; +} + +/********************************************************************* + * Driver initialization and exit + *********************************************************************/ + +static struct spi_driver snd_intelmad_driver = { + .driver = { + .name = DRIVER_NAME, + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = snd_intelmad_probe, + .remove = __devexit_p(snd_intelmad_remove), +}; + +/* +* alsa_card_intelmad_init- driver init function +* This function is called when driver module is inserted +*/ +static int __init alsa_card_intelmad_init(void) +{ + sst_dbg("called\n"); + return spi_register_driver(&snd_intelmad_driver); +} + +/** +* alsa_card_intelmad_exit- driver exit function +* This function is called when driver module is removed +*/ +static void __exit alsa_card_intelmad_exit(void) +{ + sst_dbg("called\n"); + spi_unregister_driver(&snd_intelmad_driver); +} + +module_init(alsa_card_intelmad_init) +module_exit(alsa_card_intelmad_exit) + -- 1.5.4.5

6 31

[alsa-devel] [PATCH] AC97 atmel: add support for AT91(common with AVR32).
by Sedji Gaouaou 07 Sep '09

07 Sep '09

This patch add AC97 support for ATMEL AT91 boards, using the AVR32 code. It is based on Takashi git tree(sound-2.6/for-next). Regards, Sedji Signed-off-by: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> --- sound/atmel/Kconfig | 2 +- sound/atmel/ac97c.c | 364 +++++++++++++++++++++++++++++++++++++-------------- 2 files changed, 268 insertions(+), 98 deletions(-) diff --git a/sound/atmel/Kconfig b/sound/atmel/Kconfig index 6c228a9..94de43a 100644 --- a/sound/atmel/Kconfig +++ b/sound/atmel/Kconfig @@ -12,7 +12,7 @@ config SND_ATMEL_AC97C tristate "Atmel AC97 Controller (AC97C) driver" select SND_PCM select SND_AC97_CODEC - depends on DW_DMAC && AVR32 + depends on (DW_DMAC && AVR32) || ARCH_AT91 help ALSA sound driver for the Atmel AC97 controller. diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 0c0f877..379d7ea 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -13,6 +13,7 @@ #include <linux/device.h> #include <linux/dmaengine.h> #include <linux/dma-mapping.h> +#include <linux/atmel_pdc.h> #include <linux/init.h> #include <linux/interrupt.h> #include <linux/module.h> @@ -31,6 +32,10 @@ #include <linux/dw_dmac.h> +#include <mach/cpu.h> +#include <mach/hardware.h> +#include <mach/gpio.h> + #include "ac97c.h" enum { @@ -63,6 +68,7 @@ struct atmel_ac97c { u64 cur_format; unsigned int cur_rate; unsigned long flags; + int period; /* Serialize access to opened variable */ spinlock_t lock; void __iomem *regs; @@ -167,6 +173,7 @@ static int atmel_ac97c_playback_open(struct snd_pcm_substream *substream) mutex_lock(&opened_mutex); chip->opened++; runtime->hw = atmel_ac97c_hw; + chip->period = 0; if (chip->cur_rate) { runtime->hw.rate_min = chip->cur_rate; runtime->hw.rate_max = chip->cur_rate; @@ -186,6 +193,7 @@ static int atmel_ac97c_capture_open(struct snd_pcm_substream *substream) mutex_lock(&opened_mutex); chip->opened++; runtime->hw = atmel_ac97c_hw; + chip->period = 0; if (chip->cur_rate) { runtime->hw.rate_min = chip->cur_rate; runtime->hw.rate_max = chip->cur_rate; @@ -239,12 +247,14 @@ static int atmel_ac97c_playback_hw_params(struct snd_pcm_substream *substream, retval = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); - if (retval < 0) - return retval; - /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ - if (retval == 1) - if (test_and_clear_bit(DMA_TX_READY, &chip->flags)) - dw_dma_cyclic_free(chip->dma.tx_chan); + if(cpu_is_at32ap7000()) { + if (retval < 0) + return retval; + /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ + if (retval == 1) + if (test_and_clear_bit(DMA_TX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.tx_chan); + } /* Set restrictions to params. */ mutex_lock(&opened_mutex); @@ -263,12 +273,14 @@ static int atmel_ac97c_capture_hw_params(struct snd_pcm_substream *substream, retval = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); - if (retval < 0) - return retval; - /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ - if (retval == 1) - if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) - dw_dma_cyclic_free(chip->dma.rx_chan); + if(cpu_is_at32ap7000()) { + if (retval < 0) + return retval; + /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ + if (retval == 1) + if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.rx_chan); + } /* Set restrictions to params. */ mutex_lock(&opened_mutex); @@ -282,16 +294,20 @@ static int atmel_ac97c_capture_hw_params(struct snd_pcm_substream *substream, static int atmel_ac97c_playback_hw_free(struct snd_pcm_substream *substream) { struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); - if (test_and_clear_bit(DMA_TX_READY, &chip->flags)) - dw_dma_cyclic_free(chip->dma.tx_chan); + if(cpu_is_at32ap7000()) { + if (test_and_clear_bit(DMA_TX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.tx_chan); + } return snd_pcm_lib_free_pages(substream); } static int atmel_ac97c_capture_hw_free(struct snd_pcm_substream *substream) { struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); - if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) - dw_dma_cyclic_free(chip->dma.rx_chan); + if(cpu_is_at32ap7000()) { + if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.rx_chan); + } return snd_pcm_lib_free_pages(substream); } @@ -299,6 +315,7 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream) { struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; + int block_size = frames_to_bytes(runtime, runtime->period_size); unsigned long word = ac97c_readl(chip, OCA); int retval; @@ -324,7 +341,9 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream) switch (runtime->format) { case SNDRV_PCM_FORMAT_S16_LE: - word |= AC97C_CMR_CEM_LITTLE; + if(cpu_is_at32ap7000()) { + word |= AC97C_CMR_CEM_LITTLE; + } break; case SNDRV_PCM_FORMAT_S16_BE: /* fall through */ word &= ~(AC97C_CMR_CEM_LITTLE); @@ -363,9 +382,18 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream) dev_dbg(&chip->pdev->dev, "could not set rate %d Hz\n", runtime->rate); - if (!test_bit(DMA_TX_READY, &chip->flags)) - retval = atmel_ac97c_prepare_dma(chip, substream, - DMA_TO_DEVICE); + if(cpu_is_at32ap7000()) { + if (!test_bit(DMA_TX_READY, &chip->flags)) + retval = atmel_ac97c_prepare_dma(chip, substream, + DMA_TO_DEVICE); + } else { + /* Initialize and start the PDC */ + writel(runtime->dma_addr, chip->regs + ATMEL_PDC_TPR); + writel(block_size / 2, chip->regs + ATMEL_PDC_TCR); + writel(runtime->dma_addr + block_size, + chip->regs + ATMEL_PDC_TNPR); + writel(block_size / 2, chip->regs + ATMEL_PDC_TNCR); + } return retval; } @@ -374,6 +402,7 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream) { struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; + int block_size = frames_to_bytes(runtime, runtime->period_size); unsigned long word = ac97c_readl(chip, ICA); int retval; @@ -415,11 +444,15 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream) word |= AC97C_CSR_OVRUN; ac97c_writel(chip, CAMR, word); - /* Enable channel A event interrupt */ word = ac97c_readl(chip, IMR); word |= AC97C_SR_CAEVT; - ac97c_writel(chip, IER, word); + ac97c_writel(chip, IER, /*word*/AC97C_SR_CAEVT); + + /* Enable channel A event interrupt */ + /*word = ac97c_readl(chip, IMR); + word |= AC97C_SR_CAEVT; + ac97c_writel(chip, IER, word);*/ /* set variable rate if needed */ if (runtime->rate != 48000) { @@ -438,9 +471,18 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream) dev_dbg(&chip->pdev->dev, "could not set rate %d Hz\n", runtime->rate); - if (!test_bit(DMA_RX_READY, &chip->flags)) - retval = atmel_ac97c_prepare_dma(chip, substream, - DMA_FROM_DEVICE); + if(cpu_is_at32ap7000()) { + if (!test_bit(DMA_RX_READY, &chip->flags)) + retval = atmel_ac97c_prepare_dma(chip, substream, + DMA_FROM_DEVICE); + } else { + /* Initialize and start the PDC */ + writel(runtime->dma_addr, chip->regs + ATMEL_PDC_RPR); + writel(block_size / 2, chip->regs + ATMEL_PDC_RCR); + writel(runtime->dma_addr + block_size, + chip->regs + ATMEL_PDC_RNPR); + writel(block_size / 2, chip->regs + ATMEL_PDC_RNCR); + } return retval; } @@ -449,7 +491,7 @@ static int atmel_ac97c_playback_trigger(struct snd_pcm_substream *substream, int cmd) { struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); - unsigned long camr; + unsigned long camr, ptcr = 0; int retval = 0; camr = ac97c_readl(chip, CAMR); @@ -458,15 +500,23 @@ atmel_ac97c_playback_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */ case SNDRV_PCM_TRIGGER_RESUME: /* fall through */ case SNDRV_PCM_TRIGGER_START: - retval = dw_dma_cyclic_start(chip->dma.tx_chan); - if (retval) - goto out; - camr |= AC97C_CMR_CENA; + if(cpu_is_at32ap7000()) { + retval = dw_dma_cyclic_start(chip->dma.tx_chan); + if (retval) + goto out; + } else { + ptcr = ATMEL_PDC_TXTEN; + } + camr |= AC97C_CMR_CENA | AC97C_CSR_ENDTX; break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */ case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */ case SNDRV_PCM_TRIGGER_STOP: - dw_dma_cyclic_stop(chip->dma.tx_chan); + if(cpu_is_at32ap7000()) { + dw_dma_cyclic_stop(chip->dma.tx_chan); + } else { + ptcr = ATMEL_PDC_TXTDIS; + } if (chip->opened <= 1) camr &= ~AC97C_CMR_CENA; break; @@ -476,6 +526,8 @@ atmel_ac97c_playback_trigger(struct snd_pcm_substream *substream, int cmd) } ac97c_writel(chip, CAMR, camr); + if(!cpu_is_at32ap7000()) + writel(ptcr, chip->regs + ATMEL_PDC_PTCR); out: return retval; } @@ -484,7 +536,7 @@ static int atmel_ac97c_capture_trigger(struct snd_pcm_substream *substream, int cmd) { struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); - unsigned long camr; + unsigned long camr, ptcr = 0; int retval = 0; camr = ac97c_readl(chip, CAMR); @@ -493,15 +545,24 @@ atmel_ac97c_capture_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */ case SNDRV_PCM_TRIGGER_RESUME: /* fall through */ case SNDRV_PCM_TRIGGER_START: - retval = dw_dma_cyclic_start(chip->dma.rx_chan); - if (retval) - goto out; + if(cpu_is_at32ap7000()) { + retval = dw_dma_cyclic_start(chip->dma.rx_chan); + if (retval) + goto out; + } else { + ptcr = ATMEL_PDC_RXTEN; + } camr |= AC97C_CMR_CENA; break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */ case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */ case SNDRV_PCM_TRIGGER_STOP: - dw_dma_cyclic_stop(chip->dma.rx_chan); + if(cpu_is_at32ap7000()) { + dw_dma_cyclic_stop(chip->dma.rx_chan); + } + else { + ptcr = ATMEL_PDC_RXTDIS; + } if (chip->opened <= 1) camr &= ~AC97C_CMR_CENA; break; @@ -511,6 +572,8 @@ atmel_ac97c_capture_trigger(struct snd_pcm_substream *substream, int cmd) } ac97c_writel(chip, CAMR, camr); + if(!cpu_is_at32ap7000()) + writel(ptcr, chip->regs + ATMEL_PDC_PTCR); out: return retval; } @@ -523,7 +586,11 @@ atmel_ac97c_playback_pointer(struct snd_pcm_substream *substream) snd_pcm_uframes_t frames; unsigned long bytes; - bytes = dw_dma_get_src_addr(chip->dma.tx_chan); + if(cpu_is_at32ap7000()) { + bytes = dw_dma_get_src_addr(chip->dma.tx_chan); + } else { + bytes = readl(chip->regs + ATMEL_PDC_TPR); + } bytes -= runtime->dma_addr; frames = bytes_to_frames(runtime, bytes); @@ -540,7 +607,11 @@ atmel_ac97c_capture_pointer(struct snd_pcm_substream *substream) snd_pcm_uframes_t frames; unsigned long bytes; - bytes = dw_dma_get_dst_addr(chip->dma.rx_chan); + if(cpu_is_at32ap7000()) { + bytes = dw_dma_get_dst_addr(chip->dma.rx_chan); + } else { + bytes = readl(chip->regs + ATMEL_PDC_RPR); + } bytes -= runtime->dma_addr; frames = bytes_to_frames(runtime, bytes); @@ -578,20 +649,66 @@ static irqreturn_t atmel_ac97c_interrupt(int irq, void *dev) u32 sr = ac97c_readl(chip, SR); u32 casr = ac97c_readl(chip, CASR); u32 cosr = ac97c_readl(chip, COSR); + u32 camr = ac97c_readl(chip, CAMR); if (sr & AC97C_SR_CAEVT) { - dev_info(&chip->pdev->dev, "channel A event%s%s%s%s%s%s\n", + struct snd_pcm_runtime *runtime; + int offset, next_period, block_size; + dev_dbg(&chip->pdev->dev, "channel A event%s%s%s%s%s%s\n", casr & AC97C_CSR_OVRUN ? " OVRUN" : "", casr & AC97C_CSR_RXRDY ? " RXRDY" : "", casr & AC97C_CSR_UNRUN ? " UNRUN" : "", casr & AC97C_CSR_TXEMPTY ? " TXEMPTY" : "", casr & AC97C_CSR_TXRDY ? " TXRDY" : "", !casr ? " NONE" : ""); + if(!cpu_is_at32ap7000()) { + if ((casr & camr) & AC97C_CSR_ENDTX) { + runtime = chip->playback_substream->runtime; + block_size = frames_to_bytes(runtime, + runtime->period_size); + chip->period++; + + if (chip->period == runtime->periods) + chip->period = 0; + next_period = chip->period + 1; + if (next_period == runtime->periods) + next_period = 0; + + offset = block_size * next_period; + + writel(runtime->dma_addr + offset, + chip->regs + ATMEL_PDC_TNPR); + writel(block_size / 2, + chip->regs + ATMEL_PDC_TNCR); + + snd_pcm_period_elapsed(chip->playback_substream); + } + if ((casr & camr) & AC97C_CSR_ENDRX) { + runtime = chip->capture_substream->runtime; + block_size = frames_to_bytes(runtime, + runtime->period_size); + chip->period++; + + if (chip->period == runtime->periods) + chip->period = 0; + next_period = chip->period + 1; + if (next_period == runtime->periods) + next_period = 0; + + offset = block_size * next_period; + + writel(runtime->dma_addr + offset, + chip->regs + ATMEL_PDC_RNPR); + writel(block_size / 2, + chip->regs + ATMEL_PDC_RNCR); + snd_pcm_period_elapsed(chip->capture_substream); + } + } retval = IRQ_HANDLED; } if (sr & AC97C_SR_COEVT) { - dev_info(&chip->pdev->dev, "codec channel event%s%s%s%s%s\n", + dev_dbg(&chip->pdev->dev, "codec channel event%s%s%s%s%s\n", cosr & AC97C_CSR_OVRUN ? " OVRUN" : "", cosr & AC97C_CSR_RXRDY ? " RXRDY" : "", cosr & AC97C_CSR_TXEMPTY ? " TXEMPTY" : "", @@ -608,15 +725,50 @@ static irqreturn_t atmel_ac97c_interrupt(int irq, void *dev) return retval; } +static struct ac97_pcm at91_ac97_pcm_defs[] __devinitdata = { + /* Playback */ + { + .exclusive = 1, + .r = { { + .slots = ((1 << AC97_SLOT_PCM_LEFT) + | (1 << AC97_SLOT_PCM_RIGHT)), + } }, + }, + /* PCM in */ + { + .stream = 1, + .exclusive = 1, + .r = { { + .slots = ((1 << AC97_SLOT_PCM_LEFT) + | (1 << AC97_SLOT_PCM_RIGHT)), + } } + }, + /* Mic in */ + { + .stream = 1, + .exclusive = 1, + .r = { { + .slots = (1<<AC97_SLOT_MIC), + } } + }, +}; + static int __devinit atmel_ac97c_pcm_new(struct atmel_ac97c *chip) { struct snd_pcm *pcm; struct snd_pcm_hardware hw = atmel_ac97c_hw; - int capture, playback, retval; + int capture, playback, retval, err; capture = test_bit(DMA_RX_CHAN_PRESENT, &chip->flags); playback = test_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + if(!cpu_is_at32ap7000()) { + err = snd_ac97_pcm_assign(chip->ac97_bus, + ARRAY_SIZE(at91_ac97_pcm_defs), + at91_ac97_pcm_defs); + if (err) + return err; + } retval = snd_pcm_new(chip->card, chip->card->shortname, chip->pdev->id, playback, capture, &pcm); if (retval) @@ -775,7 +927,12 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) return -ENXIO; } - pclk = clk_get(&pdev->dev, "pclk"); + if(cpu_is_at32ap7000()) { + pclk = clk_get(&pdev->dev, "pclk"); + } else { + pclk = clk_get(&pdev->dev, "ac97_clk"); + } + if (IS_ERR(pclk)) { dev_dbg(&pdev->dev, "no peripheral clock\n"); return PTR_ERR(pclk); @@ -844,44 +1001,51 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) goto err_ac97_bus; } - if (pdata->rx_dws.dma_dev) { - struct dw_dma_slave *dws = &pdata->rx_dws; - dma_cap_mask_t mask; + if(cpu_is_at32ap7000()) { + if (pdata->rx_dws.dma_dev) { + struct dw_dma_slave *dws = &pdata->rx_dws; + dma_cap_mask_t mask; - dws->rx_reg = regs->start + AC97C_CARHR + 2; + dws->rx_reg = regs->start + AC97C_CARHR + 2; - dma_cap_zero(mask); - dma_cap_set(DMA_SLAVE, mask); + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); - chip->dma.rx_chan = dma_request_channel(mask, filter, dws); + chip->dma.rx_chan = dma_request_channel(mask, filter, dws); - dev_info(&chip->pdev->dev, "using %s for DMA RX\n", + dev_info(&chip->pdev->dev, "using %s for DMA RX\n", dev_name(&chip->dma.rx_chan->dev->device)); - set_bit(DMA_RX_CHAN_PRESENT, &chip->flags); - } + set_bit(DMA_RX_CHAN_PRESENT, &chip->flags); + } - if (pdata->tx_dws.dma_dev) { - struct dw_dma_slave *dws = &pdata->tx_dws; - dma_cap_mask_t mask; + if (pdata->tx_dws.dma_dev) { + struct dw_dma_slave *dws = &pdata->tx_dws; + dma_cap_mask_t mask; - dws->tx_reg = regs->start + AC97C_CATHR + 2; + dws->tx_reg = regs->start + AC97C_CATHR + 2; - dma_cap_zero(mask); - dma_cap_set(DMA_SLAVE, mask); + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); - chip->dma.tx_chan = dma_request_channel(mask, filter, dws); + chip->dma.tx_chan = dma_request_channel(mask, filter, dws); - dev_info(&chip->pdev->dev, "using %s for DMA TX\n", - dev_name(&chip->dma.tx_chan->dev->device)); - set_bit(DMA_TX_CHAN_PRESENT, &chip->flags); - } + dev_info(&chip->pdev->dev, "using %s for DMA TX\n", + dev_name(&chip->dma.tx_chan->dev->device)); + set_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + } - if (!test_bit(DMA_RX_CHAN_PRESENT, &chip->flags) && - !test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) { - dev_dbg(&pdev->dev, "DMA not available\n"); - retval = -ENODEV; - goto err_dma; - } + if (!test_bit(DMA_RX_CHAN_PRESENT, &chip->flags) && + !test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) { + dev_dbg(&pdev->dev, "DMA not available\n"); + retval = -ENODEV; + goto err_dma; + } + } else { + /* Just pretend that we have DMA channel(for at91 i is actually + * the PDC */ + set_bit(DMA_RX_CHAN_PRESENT, &chip->flags); + set_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + } retval = atmel_ac97c_pcm_new(chip); if (retval) { @@ -897,20 +1061,22 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) platform_set_drvdata(pdev, card); - dev_info(&pdev->dev, "Atmel AC97 controller at 0x%p\n", - chip->regs); + dev_info(&pdev->dev, "Atmel AC97 controller at 0x%p, irq = %d\n", + chip->regs, irq); return 0; err_dma: - if (test_bit(DMA_RX_CHAN_PRESENT, &chip->flags)) - dma_release_channel(chip->dma.rx_chan); - if (test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) - dma_release_channel(chip->dma.tx_chan); - clear_bit(DMA_RX_CHAN_PRESENT, &chip->flags); - clear_bit(DMA_TX_CHAN_PRESENT, &chip->flags); - chip->dma.rx_chan = NULL; - chip->dma.tx_chan = NULL; + if(cpu_is_at32ap7000()) { + if (test_bit(DMA_RX_CHAN_PRESENT, &chip->flags)) + dma_release_channel(chip->dma.rx_chan); + if (test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) + dma_release_channel(chip->dma.tx_chan); + clear_bit(DMA_RX_CHAN_PRESENT, &chip->flags); + clear_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + chip->dma.rx_chan = NULL; + chip->dma.tx_chan = NULL; + } err_ac97_bus: snd_card_set_dev(card, NULL); @@ -934,12 +1100,13 @@ static int atmel_ac97c_suspend(struct platform_device *pdev, pm_message_t msg) struct snd_card *card = platform_get_drvdata(pdev); struct atmel_ac97c *chip = card->private_data; - if (test_bit(DMA_RX_READY, &chip->flags)) - dw_dma_cyclic_stop(chip->dma.rx_chan); - if (test_bit(DMA_TX_READY, &chip->flags)) - dw_dma_cyclic_stop(chip->dma.tx_chan); + if(cpu_is_at32ap7000()) { + if (test_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_stop(chip->dma.rx_chan); + if (test_bit(DMA_TX_READY, &chip->flags)) + dw_dma_cyclic_stop(chip->dma.tx_chan); + } clk_disable(chip->pclk); - return 0; } @@ -949,11 +1116,12 @@ static int atmel_ac97c_resume(struct platform_device *pdev) struct atmel_ac97c *chip = card->private_data; clk_enable(chip->pclk); - if (test_bit(DMA_RX_READY, &chip->flags)) - dw_dma_cyclic_start(chip->dma.rx_chan); - if (test_bit(DMA_TX_READY, &chip->flags)) - dw_dma_cyclic_start(chip->dma.tx_chan); - + if(cpu_is_at32ap7000()) { + if (test_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_start(chip->dma.rx_chan); + if (test_bit(DMA_TX_READY, &chip->flags)) + dw_dma_cyclic_start(chip->dma.tx_chan); + } return 0; } #else @@ -978,14 +1146,16 @@ static int __devexit atmel_ac97c_remove(struct platform_device *pdev) iounmap(chip->regs); free_irq(chip->irq, chip); - if (test_bit(DMA_RX_CHAN_PRESENT, &chip->flags)) - dma_release_channel(chip->dma.rx_chan); - if (test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) - dma_release_channel(chip->dma.tx_chan); - clear_bit(DMA_RX_CHAN_PRESENT, &chip->flags); - clear_bit(DMA_TX_CHAN_PRESENT, &chip->flags); - chip->dma.rx_chan = NULL; - chip->dma.tx_chan = NULL; + if(cpu_is_at32ap7000()) { + if (test_bit(DMA_RX_CHAN_PRESENT, &chip->flags)) + dma_release_channel(chip->dma.rx_chan); + if (test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) + dma_release_channel(chip->dma.tx_chan); + clear_bit(DMA_RX_CHAN_PRESENT, &chip->flags); + clear_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + chip->dma.rx_chan = NULL; + chip->dma.tx_chan = NULL; + } snd_card_set_dev(card, NULL); snd_card_free(card); -- 1.5.3.7

2 9

[alsa-devel] a11y: howto reduce latency with alsa's dmix plugin?
by Halim Sahin 04 Sep '09

04 Sep '09

Hello, Is there a way to reduce latency on audiocard if the dmix plugin is in use? I am asking because there is a strange problem with asistive technologies which are outputing through alsa. E. G. speech-dispatcher takes almost a half sec to stop the speech-synthesizer. If the user navigates quickly through menus, there are simultane outputs at a time. (tested with an usb logitech headset, snd-intel8x0 based card, snd-hda-intel). The same problem can be reproduced by running mplayer with alsa output driver and pressing quickly left/right arrow. You should hear several samples at a time. Using an sblive card which doesn't use dmix doesn't have such problems. Using the dmix plugin should audio much faster to avoid this problem. Please help!!! Regards Halim

2 8

[alsa-devel] [PATCH v2 1/2] Davinci: DM365: Add platform device for McBSP
by miguel.aguilar＠ridgerun.com 02 Sep '09

02 Sep '09

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> 1) Registers the platform device for McBSP on dm365. 2) Add platform data to DM365 EVM board file. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- arch/arm/mach-davinci/board-dm365-evm.c | 4 ++ arch/arm/mach-davinci/dm365.c | 45 +++++++++++++++++++++++++++- arch/arm/mach-davinci/include/mach/asp.h | 3 ++ arch/arm/mach-davinci/include/mach/dm365.h | 2 + 4 files changed, 53 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-davinci/board-dm365-evm.c b/arch/arm/mach-davinci/board-dm365-evm.c index f6adf79..fd2db78 100644 --- a/arch/arm/mach-davinci/board-dm365-evm.c +++ b/arch/arm/mach-davinci/board-dm365-evm.c @@ -177,6 +177,8 @@ static struct at24_platform_data eeprom_info = { .context = (void *)0x7f00, }; +static struct snd_platform_data dm365_evm_snd_data; + static struct i2c_board_info i2c_info[] = { { I2C_BOARD_INFO("dm365evm_keys", 0x25), @@ -476,6 +478,8 @@ static __init void dm365_evm_init(void) /* maybe setup mmc1/etc ... _after_ mmc0 */ evm_init_cpld(); + + dm365_init_asp(&dm365_evm_snd_data); } static __init void dm365_evm_irq_init(void) diff --git a/arch/arm/mach-davinci/dm365.c b/arch/arm/mach-davinci/dm365.c index e815174..c8bff14 100644 --- a/arch/arm/mach-davinci/dm365.c +++ b/arch/arm/mach-davinci/dm365.c @@ -32,6 +32,7 @@ #include <mach/time.h> #include <mach/serial.h> #include <mach/common.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -456,7 +457,7 @@ static struct davinci_clk dm365_clks[] = { CLK(NULL, "usb", &usb_clk), CLK("davinci_emac.1", NULL, &emac_clk), CLK("voice_codec", NULL, &voicecodec_clk), - CLK("soc-audio.0", NULL, &asp0_clk), + CLK("davinci-asp.0", NULL, &asp0_clk), CLK(NULL, "rto", &rto_clk), CLK(NULL, "mjcp", &mjcp_clk), CLK(NULL, NULL, NULL), @@ -603,6 +604,9 @@ INT_CFG(DM365, INT_IMX1_ENABLE, 24, 1, 1, false) INT_CFG(DM365, INT_IMX1_DISABLE, 24, 1, 0, false) INT_CFG(DM365, INT_NSF_ENABLE, 25, 1, 1, false) INT_CFG(DM365, INT_NSF_DISABLE, 25, 1, 0, false) + +EVT_CFG(DM365, EVT2_ASP_TX, 0, 1, 0, false) +EVT_CFG(DM365, EVT3_ASP_RX, 1, 1, 0, false) #endif }; @@ -806,6 +810,31 @@ static struct platform_device dm365_edma_device = { .resource = edma_resources, }; +static struct resource dm365_asp_resources[] = { + { + .start = DAVINCI_DM365_ASP0_BASE, + .end = DAVINCI_DM365_ASP0_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, + { + .start = DAVINCI_DMA_ASP0_TX, + .end = DAVINCI_DMA_ASP0_TX, + .flags = IORESOURCE_DMA, + }, + { + .start = DAVINCI_DMA_ASP0_RX, + .end = DAVINCI_DMA_ASP0_RX, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device dm365_asp_device = { + .name = "davinci-asp", + .id = 0, + .num_resources = ARRAY_SIZE(dm365_asp_resources), + .resource = dm365_asp_resources, +}; + static struct map_desc dm365_io_desc[] = { { .virtual = IO_VIRT, @@ -907,6 +936,20 @@ static struct davinci_soc_info davinci_soc_info_dm365 = { .sram_len = SZ_32K, }; +void __init dm365_init_asp(struct snd_platform_data *pdata) +{ + davinci_cfg_reg(DM365_MCBSP0_BDX); + davinci_cfg_reg(DM365_MCBSP0_X); + davinci_cfg_reg(DM365_MCBSP0_BFSX); + davinci_cfg_reg(DM365_MCBSP0_BDR); + davinci_cfg_reg(DM365_MCBSP0_R); + davinci_cfg_reg(DM365_MCBSP0_BFSR); + davinci_cfg_reg(DM365_EVT2_ASP_TX); + davinci_cfg_reg(DM365_EVT3_ASP_RX); + dm365_asp_device.dev.platform_data = pdata; + platform_device_register(&dm365_asp_device); +} + void __init dm365_init(void) { davinci_common_init(&davinci_soc_info_dm365); diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 18e4ce3..fbcbed0 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 dm365 register banks */ +#define DAVINCI_DM365_ASP0_BASE 0x01D02000 + /* Bases of da850/da830 McASP0 register banks */ #define DAVINCI_DA8XX_MCASP0_REG_BASE 0x01D00000 diff --git a/arch/arm/mach-davinci/include/mach/dm365.h b/arch/arm/mach-davinci/include/mach/dm365.h index 09db434..2291c0d 100644 --- a/arch/arm/mach-davinci/include/mach/dm365.h +++ b/arch/arm/mach-davinci/include/mach/dm365.h @@ -16,6 +16,7 @@ #include <linux/platform_device.h> #include <mach/hardware.h> #include <mach/emac.h> +#include <mach/asp.h> #define DM365_EMAC_BASE (0x01D07000) #define DM365_EMAC_CNTRL_OFFSET (0x0000) @@ -25,5 +26,6 @@ #define DM365_EMAC_CNTRL_RAM_SIZE (0x2000) void __init dm365_init(void); +void __init dm365_init_asp(struct snd_platform_data *pdata); #endif /* __ASM_ARCH_DM365_H */ -- 1.6.0.4

4 5

Re: [alsa-devel] Testers wanted: New SB X-Fi driver
by Christian Esken 01 Sep '09

01 Sep '09

On Wednesday 05 August 2009 23:21:53 Christian Esken wrote: > > 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, 2.6.30 (works) > > Kubuntu Karmic Alpha-2 64 Bit, 2.6.30 (works) > > SuSE 11.1 64 Bit, 2.6.27 (does not work) > > SuSE 11.1 64 Bit, 2.6.27 XEN (does not work) > > SuSE 11.2 Milestone3 64Bit, 2.6.30 (test canceled, due to unrelated > > problems) => I'll see whether I can redo the SuSE11.2 tests, after doing > > a bugreport at openSuSE. OK, that last issue was solved, and I now redid the test: SuSE 11.2 Milestone6 64Bit, 2.6.31-rc6 (works) I succesfully tested all analog (5.1) channels, using: speaker-test -D surround51 -c 6 I used ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa- driver-snapshot.tar.bz2 from 24.08.2009. I am just wondering, whether the new ALSA driver is it still planned for the 2.6.31 kernel?!? Christian

3 4

[alsa-devel] How to support multiple audio codecs on a single board
by Aggarwal, Anuj 01 Sep '09

01 Sep '09

Hi, I have 3 instances of AIC23 on my board and all are connected to different I2C/McBSP and hence can be used independently. How should I use/modify the current AIC23 driver and write the asoc board-specific file to enable all the AIC23 instances? Is there any similar implementation in the audio subsystem which I can look as a reference? Regards, Anuj Aggarwal

4 6

[alsa-devel] [PATCH] ALSA: allocation may fail in snd_pcm_oss_change_params()
by Roel Kluin 31 Aug '09

31 Aug '09

Allocation may fail, show if it did. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index dbe406b..ea91a0a 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1047,6 +1047,10 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream) runtime->oss.prepare = 1; vfree(runtime->oss.buffer); runtime->oss.buffer = vmalloc(runtime->oss.period_bytes); + if (runtime->oss.buffer == NULL) { + err = -ENOMEM; + goto failure; + } runtime->oss.buffer_used = 0; if (runtime->dma_area) snd_pcm_format_set_silence(runtime->format, runtime->dma_area, bytes_to_samples(runtime, runtime->dma_bytes));

2 1

[alsa-devel] [PATCH V2 2/3] add set_tdm_slot in tdm dai to define the relationship between audio channel and tdm slot
by Barry Song 31 Aug '09

31 Aug '09

For ad1938, the audio channel n just uses slot n, but for ad1836, it's different, channel 0 uses slot 0, channel 1 uses slot 4, channels 2 uses slot1, ... So add set_tdm_slot entry and use the mask field to define the relationship between audio channel and slot number. Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- sound/soc/blackfin/bf5xx-ad1938.c | 7 ++++++- sound/soc/blackfin/bf5xx-tdm-pcm.c | 12 +++++++++--- sound/soc/blackfin/bf5xx-tdm.c | 29 +++++++++++++++++++++-------- sound/soc/blackfin/bf5xx-tdm.h | 10 ++++++++++ 4 files changed, 46 insertions(+), 12 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad1938.c b/sound/soc/blackfin/bf5xx-ad1938.c index 08269e9..05163ea 100644 --- a/sound/soc/blackfin/bf5xx-ad1938.c +++ b/sound/soc/blackfin/bf5xx-ad1938.c @@ -74,8 +74,13 @@ static int bf5xx_ad1938_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + /* set cpu DAI slots, 8 channels, mask is defined as slot seq */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x76543210, 8, 32); + if (ret < 0) + return ret; + /* set codec DAI slots, 8 channels, all channels are enabled */ - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 8); + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 8, 32); if (ret < 0) return ret; diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index ccb5e82..044ef45 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -43,7 +43,7 @@ #include "bf5xx-tdm.h" #include "bf5xx-sport.h" -#define PCM_BUFFER_MAX 0x10000 +#define PCM_BUFFER_MAX 0x8000 #define FRAGMENT_SIZE_MIN (4*1024) #define FRAGMENTS_MIN 2 #define FRAGMENTS_MAX 32 @@ -177,6 +177,9 @@ out: static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, snd_pcm_uframes_t pos, void *buf, snd_pcm_uframes_t count) { + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + struct bf5xx_tdm_port *tdm_port = sport->private_data; unsigned int *src; unsigned int *dst; int i; @@ -186,9 +189,11 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, dst = (unsigned int *)substream->runtime->dma_area; dst += pos * 8; + while (count--) { for (i = 0; i < substream->runtime->channels; i++) - *(dst + i) = *src++; + *(dst + (((tdm_port->slot_seq >> + (4 * i)) & 0xF))) = *src++; dst += 8; } } else { @@ -198,7 +203,8 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, src += pos * 8; while (count--) { for (i = 0; i < substream->runtime->channels; i++) - *dst++ = *(src+i); + *dst++ = *(src + (((tdm_port->slot_seq >> + (4 * i)) & 0xF))); src += 8; } } diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c index 3096bad..0f4ee15 100644 --- a/sound/soc/blackfin/bf5xx-tdm.c +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -46,14 +46,6 @@ #include "bf5xx-sport.h" #include "bf5xx-tdm.h" -struct bf5xx_tdm_port { - u16 tcr1; - u16 rcr1; - u16 tcr2; - u16 rcr2; - int configured; -}; - static struct bf5xx_tdm_port bf5xx_tdm; static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM; @@ -181,6 +173,24 @@ static void bf5xx_tdm_shutdown(struct snd_pcm_substream *substream, bf5xx_tdm.configured = 0; } +static int bf5xx_tdm_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int mask, int slots, int width) +{ + /* + * For ad1938, the audio channel n just uses slot n, but for + * ad1836, it's different, channel 0 uses slot 0, channel 1 + * uses slot 4, channels 2 uses slot1, ... So add set_tdm_slot + * entry and use the mask field to define the relationship + * between audio channel and slot number. + * + * bit[0..3] means the slot channel 0 will use + * bit[4..7] means the slot channel 1 will use + * ... + */ + bf5xx_tdm.slot_seq = mask; + return 0; +} + #ifdef CONFIG_PM static int bf5xx_tdm_suspend(struct snd_soc_dai *dai) { @@ -235,6 +245,7 @@ static struct snd_soc_dai_ops bf5xx_tdm_dai_ops = { .hw_params = bf5xx_tdm_hw_params, .set_fmt = bf5xx_tdm_set_dai_fmt, .shutdown = bf5xx_tdm_shutdown, + .set_tdm_slot = bf5xx_tdm_set_tdm_slot, }; struct snd_soc_dai bf5xx_tdm_dai = { @@ -300,6 +311,8 @@ static int __devinit bfin_tdm_probe(struct platform_device *pdev) pr_err("Failed to register DAI: %d\n", ret); goto sport_config_err; } + + sport_handle->private_data = &bf5xx_tdm; return 0; sport_config_err: diff --git a/sound/soc/blackfin/bf5xx-tdm.h b/sound/soc/blackfin/bf5xx-tdm.h index 618ec3d..701c769 100644 --- a/sound/soc/blackfin/bf5xx-tdm.h +++ b/sound/soc/blackfin/bf5xx-tdm.h @@ -9,6 +9,16 @@ #ifndef _BF5XX_TDM_H #define _BF5XX_TDM_H +struct bf5xx_tdm_port { + u16 tcr1; + u16 rcr1; + u16 tcr2; + u16 rcr2; + /* which slot used for the corresponding audio channel? */ + int slot_seq; + int configured; +}; + extern struct snd_soc_dai bf5xx_tdm_dai; #endif -- 1.5.6.3

2 9

Re: [alsa-devel] [alsa-cvslog] alsa-tools: ALSA tools repository branch, master now at v1.0.20-5-ga539d8b
by Takashi Iwai 31 Aug '09

31 Aug '09

At Mon, 31 Aug 2009 12:32:40 +0200 (CEST), noreply-git(a)alsa-project.org wrote: > > commit a539d8b261699856095154a65e17b3b3b73155b6 > Author: Jaroslav Kysela <perex(a)perex.cz> > AuthorDate: Mon Aug 31 12:16:48 2009 +0200 > Commit: Jaroslav Kysela <perex(a)perex.cz> > CommitDate: Mon Aug 31 12:32:35 2009 +0200 > > qlo10k1: remove acinlude.m4 (QT configuration hell) - use PKG_CHECK_MODULES This breaks the build with Qt installed in the non-standard paths like /opt. Qt provides no pkgconfig files, so it can't be fixed in that way (and this program isn't for Gtk!) Please revert it. Takashi

1 0

[alsa-devel] missing control in alsamixer capture for CA0110-IBG
by Susan Cragin 31 Aug '09

31 Aug '09

I have a Creative X-fi pci card (HD-Audio Generic) that uses Creative CA0110-IBG. Under alsamixer / Capture, there used to be two adjustments / sliders. The first was for "mic" and controlled the volume. I don't remember what the second was called, but it was very useful. It's default setting was "0." But recording worked ideally if it was to one notch above "mic." I used to set them at 45/46 respectively. Anyway, this handy adjustment / slider vanished a couple of weeks ago, and I was wondering where it went? Now I only see one slider, for "mic" and the performance of my microphone isn't as good, and I have to turn it up louder.

2 1

[alsa-devel] Problems with snd-hda-intel ALC1200 on msi gt725-212us laptop Only one speaker works and headphones do not mute laptop sound
by Tony Rao 31 Aug '09

31 Aug '09

msi gt725-212us Codec: Realtek ALC1200 This laptop has 4.1 sound. Sound is only coming out of subwoofer. When I plug in headphones sounds still comes out of subwoofer, but headphones work perfectly. http://www.alsa-project.org/db/?f=e95a9401ebfa6cb0133b11fc1482ea115f766d5e Thanks, Tony R

1 0

[alsa-devel] Error enabling dmix with ALSA
by ruchi sirauthiya 30 Aug '09

30 Aug '09

Hi, I am trying to enable a software mixer (using ALSA and dmix) on an OMAP3 based embedded target board using the twl4030 codec. When I try to play audio using the command, # aplay -D playback rec.wav I am getting the following error: ALSA lib pcm_direct.c:1098:(snd_pcm_direct_initialize_poll_fd) unable to open timer 'hw:CLASS=3,SCLASS=0,CARD=0,DEV=0,SUBDEV=0' ALSA lib pcm_dmix.c:943:(snd_pcm_dmix_open) unable to initialize poll_fd aplay: main:546: audio open error: No such file or directory The contents of my /etc/asound.conf is given below. pcm.playback { type dmix ipc_key 2048 ipc_key_add_uid true ipc_gid 1 slave { rate 44100 pcm hw:0 } } I have tested it on two kernel versions, 2.6.22 and 2.6.29, but it fails on both. Playback and recording works fine if -D option is not provided. I also tried configuring ALSA (dmix) with a different configuration file which is given in the following link: http://amarok.kde.org/wiki/Setting_up_Dmix_for_ALSA # aplay -D plug:ossmix /mnt/Audio1_1.wav ALSA lib pcm_direct.c:1098:(snd_pcm_direct_initialize_poll_fd) unable to open timer 'hw:CLASS=3,SCLASS=0,CARD=0,DEV=0,SUBDEV=0' ALSA lib pcm_dmix.c:943:(snd_pcm_dmix_open) unable to initialize poll_fd aplay: main:546: audio open error: No such file or directory But as you can see the same error turns up again. The topic discussed in the following link (a similar issue) suggests that it is unable to open a second timer. http://osdir.com/ml/linux.distributions.crux.general/2005-11/msg00134.html What could be the cause of this error? Is the above error related to my ALSA configuration (for dmix) or is it related to the sound hardware (or driver)? Do I need to patch the kernel or ALSA for using dmix? Please help me resolve this issue. Let me know if any more setup info is required. Thanks and Regards, Ruchi Sirauthiya. Some information I obtained from the proc filesystem is given below for your reference. ----------------------- For Linux kernel 2.6.29 ----------------------- [root@OMAP3 tests]# cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.19 [root@OMAP3 tests]# cat /proc/asound/cards 0 [omap3evm ]: twl4030 - omap3evm omap3evm (twl4030) [root@OMAP3 tests]# omap-dss: Could not find exact pixel clock. Requested 25188 kHz, got 25187 kHz [root@OMAP3 tests]# cat /proc/asound/devices 2: : timer 3: [ 0- 0]: digital audio playback 4: [ 0- 0]: digital audio capture 5: [ 0] : control [root@OMAP3 tests]# cat /proc/asound/timers G0: system timer : 7812.500us (10000000 ticks) P0-0-0: PCM playback 0-0-0 : SLAVE P0-0-1: PCM capture 0-0-1 : SLAVE [root@OMAP3 tests]# cat /proc/asound/pcm 00-00: TWL4030 twl4030-0 : : playback 1 : capture 1 [root@OMAP3 tests]# cat /proc/asound/devices 2: : timer 3: [ 0- 0]: digital audio playback 4: [ 0- 0]: digital audio capture 5: [ 0] : control [root@OMAP3 tests]# cat /proc/asound/pcm 00-00: TWL4030 twl4030-0 : : playback 1 : capture 1 [root@OMAP3 tests]# cat /proc/asound/oss/devices 3: [0- 0]: digital audio 4: [0- 0]: digital audio [root@OMAP3 tests]# cat /proc/asound/oss/sndstat Sound Driver:3.8.1a-980706 (ALSA v1.0.19 emulation code) Kernel: Linux OMAP3 2.6.29-rc3-omap1 #1 Fri Jul 24 14:45:37 IST 2009 armv7l Config options: 0 Installed drivers: Type 10: ALSA emulation Card config: omap3evm (twl4030) Audio devices: 0: (DUPLEX) Synth devices: NOT ENABLED IN CONFIG Midi devices: NOT ENABLED IN CONFIG Timers: 31: system timer Mixers: NOT ENABLED IN CONFIG [root@OMAP3 tests]# aplay -D playback /mnt/Audio1_1.wav ALSA lib pcm_direct.c:1098:(snd_pcm_direct_initialize_poll_fd) unable to open timer 'hw:CLASS=3,SCLASS=0,CARD=0,DEV=0,SUBDEV=0' ALSA lib pcm_dmix.c:943:(snd_pcm_dmix_open) unable to initialize poll_fd aplay: main:546: audio open error: No such file or directory ----------------------- For Linux kernel 2.6.22 ----------------------- [root@OMAP3 tests]# cat /proc/asound/devices 2: : timer 3: [ 0- 0]: digital audio playback 4: [ 0- 0]: digital audio capture 5: [ 0] : control [root@OMAP3 tests]# cat /proc/asound/pcm 00-00: OMAP PCM : omap alsa pcm : playback 1 : capture 1 [root@OMAP3 tests]# cat /proc/asound/timers G0: system timer : 7812.500us (10000000 ticks) P0-0-0: PCM playback 0-0-0 : SLAVE P0-0-1: PCM capture 0-0-1 : SLAVE [root@OMAP3 tests]# cat /proc/asound/ /proc/asound/TWL4030/ /proc/asound/devices /proc/asound/timers /proc/asound/card0/ /proc/asound/hwdep /proc/asound/version /proc/asound/cards /proc/asound/pcm [root@OMAP3 tests]# cat /proc/asound/cards 0 [TWL4030 ]: OMAP_ALSA - TWL4030 TWL4030 /proc/asound/oss was not found in the proc file system when booted with kernel 2.6.22.

2 4

[alsa-devel] ASoC DAI on Atmel AT32AP7000
by Christos Nikolaou 30 Aug '09

30 Aug '09

Hi, Up to 2.6.28 the ASoC framework was working on AT32AP7000 Atmel platform. Now I try to recompile it on 2.6.31-rc8 and I get Advance Linux Sound Architecture Driver Version 1.0.20. No device for DAI atmel-ssc0 No device for DAI atmel-ssc1 No device for DAI atmel-ssc2 whereas the ssc devices are properly (as in 2.6.28) initialized ssc ssc.0: Atmel SSC device at 0xffe1c00 (irq 10) ssc ssc.1: Atmel SSC device at 0xffe2000 (irq 11) ssc ssc.2: Atmel SSC device at 0xffe2400 (irq 12) It looks that this message comes from snd_soc_register_dai in soc-core.c if (!dai->dev) printk(KERN_WARNING "No device for DAI %s\n", dai->name); it seems for some reason it thinks there is no device in dai->dev Unless I'm mistaken, that is being called from sound/soc/atmel/atmel_ssc_dai.c after some modifications done by broonie on these files according to http://git.kernel.org/?p=linux/kernel/git/torvalds/linux-2.6.git;a=history;… . Is there anything that can be done to rectify this? Regards, Christos

2 3

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 30 Aug '09

30 Aug '09

The following changes since commit a0a499c5792b8656cd51e11d5e0db9fb21640f58: Eduardo Valentin (1): ASoC: OMAP: Use DMA operating mode of McBSP are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Ben Dooks (4): ASoC: S3C24XX: Add audio core and tlv320aic23 for Simtec boards ASoC: tlv320aic3x: Change to use device model ASoC: tlv320aic3x: fixup board device changes ASoC: S3C24XX: Support for Simtec Hermes boards Kuninori Morimoto (2): ASoC: Add ak4642/ak4643 codec support ASoC: Add FSI-AK4642 sound support for SuperH Marek Vasut (5): power_supply: Make wm97xx-core pass platform_data to battery, ts power_supply: Convert WM97xx battery to use platform_data power_supply: Let wm97xx-battery use irq to detect charger state [ARM] pxa: Convert all platforms to pass platform_data to wm97xx battery ASoC: Pass correct platform data from pxa2xx-ac97 Mark Brown (1): ASoC: Add DAPM widget power decision debugfs files Roel Kluin (1): ASoC: free socdev if init_card() fails in wm9705_soc_probe() arch/arm/mach-pxa/mioa701.c | 10 +- arch/arm/mach-pxa/palmld.c | 19 +- arch/arm/mach-pxa/palmt5.c | 19 +- arch/arm/mach-pxa/palmte2.c | 20 +- arch/arm/mach-pxa/palmtx.c | 19 +- arch/arm/mach-pxa/palmz72.c | 20 +- arch/arm/mach-pxa/treo680.c | 1 - arch/arm/plat-s3c/include/plat/audio-simtec.h | 37 ++ drivers/input/touchscreen/wm97xx-core.c | 3 + drivers/power/wm97xx_battery.c | 56 ++- include/linux/wm97xx.h | 18 + include/linux/wm97xx_batt.h | 26 -- include/sound/soc-dapm.h | 1 + include/sound/soc.h | 1 + sound/arm/pxa2xx-ac97.c | 4 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4642.c | 502 ++++++++++++++++++++++++ sound/soc/codecs/ak4642.h | 20 + sound/soc/codecs/tlv320aic3x.c | 221 ++++++----- sound/soc/codecs/tlv320aic3x.h | 2 - sound/soc/codecs/wm9705.c | 2 +- sound/soc/davinci/davinci-evm.c | 12 +- sound/soc/omap/n810.c | 12 +- sound/soc/pxa/pxa2xx-ac97.c | 4 +- sound/soc/s3c24xx/Kconfig | 19 + sound/soc/s3c24xx/Makefile | 7 + sound/soc/s3c24xx/s3c24xx_simtec.c | 394 +++++++++++++++++++ sound/soc/s3c24xx/s3c24xx_simtec.h | 22 + sound/soc/s3c24xx/s3c24xx_simtec_hermes.c | 153 +++++++ sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c | 137 +++++++ sound/soc/s6000/s6105-ipcam.c | 12 +- sound/soc/sh/Kconfig | 8 + sound/soc/sh/Makefile | 2 + sound/soc/sh/fsi-ak4642.c | 107 +++++ sound/soc/soc-core.c | 8 + sound/soc/soc-dapm.c | 86 ++++ 37 files changed, 1784 insertions(+), 206 deletions(-) create mode 100644 arch/arm/plat-s3c/include/plat/audio-simtec.h delete mode 100644 include/linux/wm97xx_batt.h create mode 100644 sound/soc/codecs/ak4642.c create mode 100644 sound/soc/codecs/ak4642.h create mode 100644 sound/soc/s3c24xx/s3c24xx_simtec.c create mode 100644 sound/soc/s3c24xx/s3c24xx_simtec.h create mode 100644 sound/soc/s3c24xx/s3c24xx_simtec_hermes.c create mode 100644 sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c create mode 100644 sound/soc/sh/fsi-ak4642.c

3 8

[alsa-devel] [PATCH 4/4] ASoC: Davinci: Add audio codec support for DM365 EVM
by miguel.aguilar＠ridgerun.com 29 Aug '09

29 Aug '09

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> 1) Enables tlv320aic3101 support on DM365 EVM. 2) Set tlv320aic3x i2c setup into DM365 EVM board file. This patch was tested on DM365 EVM rev c. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- arch/arm/mach-davinci/board-dm365-evm.c | 3 +++ sound/soc/davinci/Kconfig | 9 +++++++++ sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 29 +++++++++++++++++++++++++++-- 4 files changed, 40 insertions(+), 2 deletions(-) diff --git a/arch/arm/mach-davinci/board-dm365-evm.c b/arch/arm/mach-davinci/board-dm365-evm.c index fd2db78..cff68e1 100644 --- a/arch/arm/mach-davinci/board-dm365-evm.c +++ b/arch/arm/mach-davinci/board-dm365-evm.c @@ -187,6 +187,9 @@ static struct i2c_board_info i2c_info[] = { I2C_BOARD_INFO("24c256", 0x50), .platform_data = &eeprom_info, }, + { + I2C_BOARD_INFO("tlv320aic3x", 0x18), + }, }; static struct davinci_i2c_platform_data i2c_pdata = { diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 6802dd5..b78b882 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -32,6 +32,15 @@ config SND_DM6467_SOC_EVM help Say Y if you want to add support for SoC audio on TI +config SND_DM365_SOC_EVM + tristate "SoC Audio support for DaVinci DM365 EVM" + depends on SND_DAVINCI_SOC && 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 + config SND_DAVINCI_SOC_SFFSDR tristate "SoC Audio support for SFFSDR" depends on SND_DAVINCI_SOC && MACH_SFFSDR diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index 67be54f..8238612 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -13,4 +13,5 @@ snd-soc-sffsdr-objs := davinci-sffsdr.o obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DM6467_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DM365_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index f3bb6f6..a859208 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -44,7 +44,8 @@ static int evm_hw_params(struct snd_pcm_substream *substream, unsigned sysclk; /* ASP1 on DM355 EVM is clocked by an external oscillator */ - if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm()) + if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm() + || machine_is_davinci_dm365_evm()) sysclk = 27000000; /* ASP0 in DM6446 EVM is clocked by U55, as configured by @@ -179,6 +180,14 @@ static struct snd_soc_card dm6467_snd_soc_card_evm = { .num_links = ARRAY_SIZE(dm6467_evm_dai), }; +/* davinci dm365 evm audio machine driver */ +static struct snd_soc_card dm365_snd_soc_card_evm = { + .name = "DaVinci DM365 EVM", + .platform = &davinci_soc_platform, + .dai_link = &evm_dai, + .num_links = 1, +}; + /* evm audio private data */ static struct aic3x_setup_data evm_aic3x_setup = { .i2c_bus = 1, @@ -191,6 +200,12 @@ static struct aic3x_setup_data dm6467_evm_aic3x_setup = { .i2c_address = 0x18, }; +/* dm365 evm audio private data */ +static struct aic3x_setup_data dm365_evm_aic3x_setup = { + .i2c_bus = 1, + .i2c_address = 0x18, +}; + /* evm audio subsystem */ static struct snd_soc_device evm_snd_devdata = { .card = &snd_soc_card_evm, @@ -198,13 +213,20 @@ static struct snd_soc_device evm_snd_devdata = { .codec_data = &evm_aic3x_setup, }; -/* evm audio subsystem */ +/* dm6467 evm audio subsystem */ static struct snd_soc_device dm6467_evm_snd_devdata = { .card = &dm6467_snd_soc_card_evm, .codec_dev = &soc_codec_dev_aic3x, .codec_data = &dm6467_evm_aic3x_setup, }; +/* dm365 evm audio subsystem */ +static struct snd_soc_device dm365_evm_snd_devdata = { + .card = &dm365_snd_soc_card_evm, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &dm365_evm_aic3x_setup, +}; + static struct platform_device *evm_snd_device; static int __init evm_init(void) @@ -222,6 +244,9 @@ static int __init evm_init(void) } else if (machine_is_davinci_dm6467_evm()) { evm_snd_dev_data = &dm6467_evm_snd_devdata; index = 0; + }else if (machine_is_davinci_dm365_evm()) { + evm_snd_dev_data = &dm365_evm_snd_devdata; + index = 0; } else return -EINVAL; -- 1.6.0.4

2 1

[alsa-devel] [PATCH 3/4] Davinci: DM365: Add platform device for McBSP
by miguel.aguilar＠ridgerun.com 29 Aug '09

29 Aug '09

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> 1) Registers the platform device for McBSP on DM365. 2) Add platform data to DM365 EVM board file. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- arch/arm/mach-davinci/board-dm365-evm.c | 4 ++ arch/arm/mach-davinci/dm365.c | 42 +++++++++++++++++++++++++++- arch/arm/mach-davinci/include/mach/asp.h | 3 ++ arch/arm/mach-davinci/include/mach/dm365.h | 2 + 4 files changed, 50 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-davinci/board-dm365-evm.c b/arch/arm/mach-davinci/board-dm365-evm.c index f6adf79..fd2db78 100644 --- a/arch/arm/mach-davinci/board-dm365-evm.c +++ b/arch/arm/mach-davinci/board-dm365-evm.c @@ -177,6 +177,8 @@ static struct at24_platform_data eeprom_info = { .context = (void *)0x7f00, }; +static struct snd_platform_data dm365_evm_snd_data; + static struct i2c_board_info i2c_info[] = { { I2C_BOARD_INFO("dm365evm_keys", 0x25), @@ -476,6 +478,8 @@ static __init void dm365_evm_init(void) /* maybe setup mmc1/etc ... _after_ mmc0 */ evm_init_cpld(); + + dm365_init_asp(&dm365_evm_snd_data); } static __init void dm365_evm_irq_init(void) diff --git a/arch/arm/mach-davinci/dm365.c b/arch/arm/mach-davinci/dm365.c index e815174..cba09b0 100644 --- a/arch/arm/mach-davinci/dm365.c +++ b/arch/arm/mach-davinci/dm365.c @@ -32,6 +32,7 @@ #include <mach/time.h> #include <mach/serial.h> #include <mach/common.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -456,7 +457,7 @@ static struct davinci_clk dm365_clks[] = { CLK(NULL, "usb", &usb_clk), CLK("davinci_emac.1", NULL, &emac_clk), CLK("voice_codec", NULL, &voicecodec_clk), - CLK("soc-audio.0", NULL, &asp0_clk), + CLK("davinci-asp.0", NULL, &asp0_clk), CLK(NULL, "rto", &rto_clk), CLK(NULL, "mjcp", &mjcp_clk), CLK(NULL, NULL, NULL), @@ -806,6 +807,31 @@ static struct platform_device dm365_edma_device = { .resource = edma_resources, }; +static struct resource dm365_asp_resources[] = { + { + .start = DAVINCI_DM365_ASP0_BASE, + .end = DAVINCI_DM365_ASP0_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, + { + .start = DAVINCI_DMA_ASP0_TX, + .end = DAVINCI_DMA_ASP0_TX, + .flags = IORESOURCE_DMA, + }, + { + .start = DAVINCI_DMA_ASP0_RX, + .end = DAVINCI_DMA_ASP0_RX, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device dm365_asp_device = { + .name = "davinci-asp", + .id = 0, + .num_resources = ARRAY_SIZE(dm365_asp_resources), + .resource = dm365_asp_resources, +}; + static struct map_desc dm365_io_desc[] = { { .virtual = IO_VIRT, @@ -907,6 +933,20 @@ static struct davinci_soc_info davinci_soc_info_dm365 = { .sram_len = SZ_32K, }; +void __init dm365_init_asp(struct snd_platform_data *pdata) +{ + davinci_cfg_reg(DM365_MCBSP0_BDX); + davinci_cfg_reg(DM365_MCBSP0_X); + davinci_cfg_reg(DM365_MCBSP0_BFSX); + davinci_cfg_reg(DM365_MCBSP0_BDR); + davinci_cfg_reg(DM365_MCBSP0_R); + davinci_cfg_reg(DM365_MCBSP0_BFSR); + davinci_cfg_reg(DM365_EVT2_ASP_TX); + davinci_cfg_reg(DM365_EVT3_ASP_RX); + dm365_asp_device.dev.platform_data = pdata; + platform_device_register(&dm365_asp_device); +} + void __init dm365_init(void) { davinci_common_init(&davinci_soc_info_dm365); diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 18e4ce3..fbcbed0 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 dm365 register banks */ +#define DAVINCI_DM365_ASP0_BASE 0x01D02000 + /* Bases of da850/da830 McASP0 register banks */ #define DAVINCI_DA8XX_MCASP0_REG_BASE 0x01D00000 diff --git a/arch/arm/mach-davinci/include/mach/dm365.h b/arch/arm/mach-davinci/include/mach/dm365.h index 09db434..2291c0d 100644 --- a/arch/arm/mach-davinci/include/mach/dm365.h +++ b/arch/arm/mach-davinci/include/mach/dm365.h @@ -16,6 +16,7 @@ #include <linux/platform_device.h> #include <mach/hardware.h> #include <mach/emac.h> +#include <mach/asp.h> #define DM365_EMAC_BASE (0x01D07000) #define DM365_EMAC_CNTRL_OFFSET (0x0000) @@ -25,5 +26,6 @@ #define DM365_EMAC_CNTRL_RAM_SIZE (0x2000) void __init dm365_init(void); +void __init dm365_init_asp(struct snd_platform_data *pdata); #endif /* __ASM_ARCH_DM365_H */ -- 1.6.0.4

2 1

[alsa-devel] [PATCH v2 2/2] ASoC: Davinci: Add audio codec support for DM365 EVM
by miguel.aguilar＠ridgerun.com 29 Aug '09

29 Aug '09

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> This patch enables tlv320aic3101 support on DM365 EVM and it was tested on DM365 EVM rev c. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- sound/soc/davinci/Kconfig | 9 +++++++++ sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 29 +++++++++++++++++++++++++++-- 3 files changed, 37 insertions(+), 2 deletions(-) diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 6802dd5..b78b882 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -32,6 +32,15 @@ config SND_DM6467_SOC_EVM help Say Y if you want to add support for SoC audio on TI +config SND_DM365_SOC_EVM + tristate "SoC Audio support for DaVinci DM365 EVM" + depends on SND_DAVINCI_SOC && 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 + config SND_DAVINCI_SOC_SFFSDR tristate "SoC Audio support for SFFSDR" depends on SND_DAVINCI_SOC && MACH_SFFSDR diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index 67be54f..8238612 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -13,4 +13,5 @@ snd-soc-sffsdr-objs := davinci-sffsdr.o obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DM6467_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DM365_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index f3bb6f6..a859208 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -44,7 +44,8 @@ static int evm_hw_params(struct snd_pcm_substream *substream, unsigned sysclk; /* ASP1 on DM355 EVM is clocked by an external oscillator */ - if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm()) + if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm() + || machine_is_davinci_dm365_evm()) sysclk = 27000000; /* ASP0 in DM6446 EVM is clocked by U55, as configured by @@ -179,6 +180,14 @@ static struct snd_soc_card dm6467_snd_soc_card_evm = { .num_links = ARRAY_SIZE(dm6467_evm_dai), }; +/* davinci dm365 evm audio machine driver */ +static struct snd_soc_card dm365_snd_soc_card_evm = { + .name = "DaVinci DM365 EVM", + .platform = &davinci_soc_platform, + .dai_link = &evm_dai, + .num_links = 1, +}; + /* evm audio private data */ static struct aic3x_setup_data evm_aic3x_setup = { .i2c_bus = 1, @@ -191,6 +200,12 @@ static struct aic3x_setup_data dm6467_evm_aic3x_setup = { .i2c_address = 0x18, }; +/* dm365 evm audio private data */ +static struct aic3x_setup_data dm365_evm_aic3x_setup = { + .i2c_bus = 1, + .i2c_address = 0x18, +}; + /* evm audio subsystem */ static struct snd_soc_device evm_snd_devdata = { .card = &snd_soc_card_evm, @@ -198,13 +213,20 @@ static struct snd_soc_device evm_snd_devdata = { .codec_data = &evm_aic3x_setup, }; -/* evm audio subsystem */ +/* dm6467 evm audio subsystem */ static struct snd_soc_device dm6467_evm_snd_devdata = { .card = &dm6467_snd_soc_card_evm, .codec_dev = &soc_codec_dev_aic3x, .codec_data = &dm6467_evm_aic3x_setup, }; +/* dm365 evm audio subsystem */ +static struct snd_soc_device dm365_evm_snd_devdata = { + .card = &dm365_snd_soc_card_evm, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &dm365_evm_aic3x_setup, +}; + static struct platform_device *evm_snd_device; static int __init evm_init(void) @@ -222,6 +244,9 @@ static int __init evm_init(void) } else if (machine_is_davinci_dm6467_evm()) { evm_snd_dev_data = &dm6467_evm_snd_devdata; index = 0; + }else if (machine_is_davinci_dm365_evm()) { + evm_snd_dev_data = &dm365_evm_snd_devdata; + index = 0; } else return -EINVAL; -- 1.6.0.4

1 0

[alsa-devel] ASoC: at91sam9260-PCM1808/TAS5709-based board driver
by Pedro Sanchez 29 Aug '09

29 Aug '09

Hello, I have to develop a driver for a custom board based on the at91sam9260 with a PCM1808/TAS5709 chipset combo. I am addressing the PCM1808 first because it is a simpler device. Since I'm new to ASoC I'm looking for some help in this list on how to go about writing this driver. So far I have identified two files that I need to work on (other than the needed kconfig modifications). First is the codec driver at sound/soc/codecs/PCM1808.[c|h] which is based on the existent pcm3008 driver; I believe I'm fine with this. The second file is the machine driver at sound/soc/atmel/myBoard.c. I'm using sam9g20_wm8731.c as reference, on which I believe I found a typo which I reported in a separated e-mail. So my first question is, is this two-files step the right approach? am I in the right track? I'm using the 2.6.30 kernel. Other questions will come and I hope I will get some cycles from you to help me direct my work. Of course I'll be glad to submit my driver work for inclusion in the kernel if it proves useful. Regards, -- Pedro

5 16

[alsa-devel] [PATCH 1/2] Davinci: DM365: Add platform device for McBSP
by miguel.aguilar＠ridgerun.com 28 Aug '09

28 Aug '09

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> 1) Registers the platform device for McBSP on DM365. 2) Add platform data to DM365 EVM board file. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- arch/arm/mach-davinci/board-dm365-evm.c | 4 ++ arch/arm/mach-davinci/dm365.c | 42 +++++++++++++++++++++++++++- arch/arm/mach-davinci/include/mach/asp.h | 3 ++ arch/arm/mach-davinci/include/mach/dm365.h | 2 + 4 files changed, 50 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-davinci/board-dm365-evm.c b/arch/arm/mach-davinci/board-dm365-evm.c index f6adf79..fd2db78 100644 --- a/arch/arm/mach-davinci/board-dm365-evm.c +++ b/arch/arm/mach-davinci/board-dm365-evm.c @@ -177,6 +177,8 @@ static struct at24_platform_data eeprom_info = { .context = (void *)0x7f00, }; +static struct snd_platform_data dm365_evm_snd_data; + static struct i2c_board_info i2c_info[] = { { I2C_BOARD_INFO("dm365evm_keys", 0x25), @@ -476,6 +478,8 @@ static __init void dm365_evm_init(void) /* maybe setup mmc1/etc ... _after_ mmc0 */ evm_init_cpld(); + + dm365_init_asp(&dm365_evm_snd_data); } static __init void dm365_evm_irq_init(void) diff --git a/arch/arm/mach-davinci/dm365.c b/arch/arm/mach-davinci/dm365.c index e815174..cba09b0 100644 --- a/arch/arm/mach-davinci/dm365.c +++ b/arch/arm/mach-davinci/dm365.c @@ -32,6 +32,7 @@ #include <mach/time.h> #include <mach/serial.h> #include <mach/common.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -456,7 +457,7 @@ static struct davinci_clk dm365_clks[] = { CLK(NULL, "usb", &usb_clk), CLK("davinci_emac.1", NULL, &emac_clk), CLK("voice_codec", NULL, &voicecodec_clk), - CLK("soc-audio.0", NULL, &asp0_clk), + CLK("davinci-asp.0", NULL, &asp0_clk), CLK(NULL, "rto", &rto_clk), CLK(NULL, "mjcp", &mjcp_clk), CLK(NULL, NULL, NULL), @@ -806,6 +807,31 @@ static struct platform_device dm365_edma_device = { .resource = edma_resources, }; +static struct resource dm365_asp_resources[] = { + { + .start = DAVINCI_DM365_ASP0_BASE, + .end = DAVINCI_DM365_ASP0_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, + { + .start = DAVINCI_DMA_ASP0_TX, + .end = DAVINCI_DMA_ASP0_TX, + .flags = IORESOURCE_DMA, + }, + { + .start = DAVINCI_DMA_ASP0_RX, + .end = DAVINCI_DMA_ASP0_RX, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device dm365_asp_device = { + .name = "davinci-asp", + .id = 0, + .num_resources = ARRAY_SIZE(dm365_asp_resources), + .resource = dm365_asp_resources, +}; + static struct map_desc dm365_io_desc[] = { { .virtual = IO_VIRT, @@ -907,6 +933,20 @@ static struct davinci_soc_info davinci_soc_info_dm365 = { .sram_len = SZ_32K, }; +void __init dm365_init_asp(struct snd_platform_data *pdata) +{ + davinci_cfg_reg(DM365_MCBSP0_BDX); + davinci_cfg_reg(DM365_MCBSP0_X); + davinci_cfg_reg(DM365_MCBSP0_BFSX); + davinci_cfg_reg(DM365_MCBSP0_BDR); + davinci_cfg_reg(DM365_MCBSP0_R); + davinci_cfg_reg(DM365_MCBSP0_BFSR); + davinci_cfg_reg(DM365_EVT2_ASP_TX); + davinci_cfg_reg(DM365_EVT3_ASP_RX); + dm365_asp_device.dev.platform_data = pdata; + platform_device_register(&dm365_asp_device); +} + void __init dm365_init(void) { davinci_common_init(&davinci_soc_info_dm365); diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 18e4ce3..fbcbed0 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 dm365 register banks */ +#define DAVINCI_DM365_ASP0_BASE 0x01D02000 + /* Bases of da850/da830 McASP0 register banks */ #define DAVINCI_DA8XX_MCASP0_REG_BASE 0x01D00000 diff --git a/arch/arm/mach-davinci/include/mach/dm365.h b/arch/arm/mach-davinci/include/mach/dm365.h index 09db434..2291c0d 100644 --- a/arch/arm/mach-davinci/include/mach/dm365.h +++ b/arch/arm/mach-davinci/include/mach/dm365.h @@ -16,6 +16,7 @@ #include <linux/platform_device.h> #include <mach/hardware.h> #include <mach/emac.h> +#include <mach/asp.h> #define DM365_EMAC_BASE (0x01D07000) #define DM365_EMAC_CNTRL_OFFSET (0x0000) @@ -25,5 +26,6 @@ #define DM365_EMAC_CNTRL_RAM_SIZE (0x2000) void __init dm365_init(void); +void __init dm365_init_asp(struct snd_platform_data *pdata); #endif /* __ASM_ARCH_DM365_H */ -- 1.6.0.4

2 1

[alsa-devel] [PATCH 2/2] ASoC: Davinci: Add audio codec support for DM365 EVM
by miguel.aguilar＠ridgerun.com 28 Aug '09

28 Aug '09

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> 1) Enables tlv320aic3101 support on DM365 EVM. 2) Set tlv320aic3x i2c setup into DM365 EVM board file. This patch was tested on DM365 EVM rev c. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- arch/arm/mach-davinci/board-dm365-evm.c | 3 +++ sound/soc/davinci/Kconfig | 9 +++++++++ sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 29 +++++++++++++++++++++++++++-- 4 files changed, 40 insertions(+), 2 deletions(-) diff --git a/arch/arm/mach-davinci/board-dm365-evm.c b/arch/arm/mach-davinci/board-dm365-evm.c index fd2db78..cff68e1 100644 --- a/arch/arm/mach-davinci/board-dm365-evm.c +++ b/arch/arm/mach-davinci/board-dm365-evm.c @@ -187,6 +187,9 @@ static struct i2c_board_info i2c_info[] = { I2C_BOARD_INFO("24c256", 0x50), .platform_data = &eeprom_info, }, + { + I2C_BOARD_INFO("tlv320aic3x", 0x18), + }, }; static struct davinci_i2c_platform_data i2c_pdata = { diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 6802dd5..b78b882 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -32,6 +32,15 @@ config SND_DM6467_SOC_EVM help Say Y if you want to add support for SoC audio on TI +config SND_DM365_SOC_EVM + tristate "SoC Audio support for DaVinci DM365 EVM" + depends on SND_DAVINCI_SOC && 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 + config SND_DAVINCI_SOC_SFFSDR tristate "SoC Audio support for SFFSDR" depends on SND_DAVINCI_SOC && MACH_SFFSDR diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index 67be54f..8238612 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -13,4 +13,5 @@ snd-soc-sffsdr-objs := davinci-sffsdr.o obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DM6467_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DM365_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index f3bb6f6..a859208 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -44,7 +44,8 @@ static int evm_hw_params(struct snd_pcm_substream *substream, unsigned sysclk; /* ASP1 on DM355 EVM is clocked by an external oscillator */ - if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm()) + if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm() + || machine_is_davinci_dm365_evm()) sysclk = 27000000; /* ASP0 in DM6446 EVM is clocked by U55, as configured by @@ -179,6 +180,14 @@ static struct snd_soc_card dm6467_snd_soc_card_evm = { .num_links = ARRAY_SIZE(dm6467_evm_dai), }; +/* davinci dm365 evm audio machine driver */ +static struct snd_soc_card dm365_snd_soc_card_evm = { + .name = "DaVinci DM365 EVM", + .platform = &davinci_soc_platform, + .dai_link = &evm_dai, + .num_links = 1, +}; + /* evm audio private data */ static struct aic3x_setup_data evm_aic3x_setup = { .i2c_bus = 1, @@ -191,6 +200,12 @@ static struct aic3x_setup_data dm6467_evm_aic3x_setup = { .i2c_address = 0x18, }; +/* dm365 evm audio private data */ +static struct aic3x_setup_data dm365_evm_aic3x_setup = { + .i2c_bus = 1, + .i2c_address = 0x18, +}; + /* evm audio subsystem */ static struct snd_soc_device evm_snd_devdata = { .card = &snd_soc_card_evm, @@ -198,13 +213,20 @@ static struct snd_soc_device evm_snd_devdata = { .codec_data = &evm_aic3x_setup, }; -/* evm audio subsystem */ +/* dm6467 evm audio subsystem */ static struct snd_soc_device dm6467_evm_snd_devdata = { .card = &dm6467_snd_soc_card_evm, .codec_dev = &soc_codec_dev_aic3x, .codec_data = &dm6467_evm_aic3x_setup, }; +/* dm365 evm audio subsystem */ +static struct snd_soc_device dm365_evm_snd_devdata = { + .card = &dm365_snd_soc_card_evm, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &dm365_evm_aic3x_setup, +}; + static struct platform_device *evm_snd_device; static int __init evm_init(void) @@ -222,6 +244,9 @@ static int __init evm_init(void) } else if (machine_is_davinci_dm6467_evm()) { evm_snd_dev_data = &dm6467_evm_snd_devdata; index = 0; + }else if (machine_is_davinci_dm365_evm()) { + evm_snd_dev_data = &dm365_evm_snd_devdata; + index = 0; } else return -EINVAL; -- 1.6.0.4

2 1

[alsa-devel] [PATCH] Davinci: Enable ASoC support for DM365 EVM in defconfig
by miguel.aguilar＠ridgerun.com 28 Aug '09

28 Aug '09

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- arch/arm/configs/davinci_all_defconfig | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/arch/arm/configs/davinci_all_defconfig b/arch/arm/configs/davinci_all_defconfig index ec63c15..cbe18b4 100644 --- a/arch/arm/configs/davinci_all_defconfig +++ b/arch/arm/configs/davinci_all_defconfig @@ -1099,6 +1099,7 @@ CONFIG_SND_DAVINCI_SOC_I2S=m CONFIG_SND_DAVINCI_SOC_MCASP=m CONFIG_SND_DAVINCI_SOC_EVM=m CONFIG_SND_DM6467_SOC_EVM=m +CONFIG_SND_DM365_SOC_EVM=m # CONFIG_SND_DAVINCI_SOC_SFFSDR is not set CONFIG_SND_SOC_I2C_AND_SPI=m # CONFIG_SND_SOC_ALL_CODECS is not set -- 1.6.0.4

1 0

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 28 Aug '09

28 Aug '09

The following changes since commit d09a2afc9359407114b7062519101f1ee2d05388: Jarkko Nikula (1): ARM: OMAP: McBSP: Merge two functions into omap_mcbsp_start/_stop are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Candelaria Villareal, Jorge (1): ASoC: SDP3430: Fix TWL GPIO6 pin mux request Mark Brown (4): ASoC: Add S3C24xx dependencies for Simtec machines ASoC: Make platform data optional for TLV320AIC3x ASoC: Fix s3c-i2s-v2 build ASoC: Don't reconfigure WM8350 FLL if not needed Shine Liu (1): ASoC: S3C platform: Fix s3c2410_dma_started() called at improper time sound/soc/codecs/tlv320aic3x.c | 13 ++++++------- sound/soc/codecs/wm8350.c | 9 +++++++++ sound/soc/omap/sdp3430.c | 15 +++++++++++---- sound/soc/s3c24xx/Kconfig | 4 ++-- sound/soc/s3c24xx/s3c-i2s-v2.c | 11 +++++++++++ sound/soc/s3c24xx/s3c2443-ac97.c | 10 ++++++++++ sound/soc/s3c24xx/s3c24xx-i2s.c | 5 +++++ sound/soc/s3c24xx/s3c24xx-pcm.c | 1 - 8 files changed, 54 insertions(+), 14 deletions(-)

2 5

[alsa-devel] Audio capture not working with AIC23/McBSP1 on OMAP3517
by Aggarwal, Anuj 28 Aug '09

28 Aug '09

Hi, I am trying to capture audio through LineIn on my OMAP3517 EVM, using AIC23 codec and McBSP1, and getting the following error: arecord: pcm_read:1529: read error: Input/output error Playback is working fine using the same combination. I have tried using another instance of AIC23 codec which uses McBSP2 and both playback & capture works fine there. Has anyone faced this issue before? Any suggestions? Regards, Anuj Aggarwal

4 11

[alsa-devel] Skype beta better audio, but crashing after launch...
by jack craig 28 Aug '09

28 Aug '09

I thought i'd start a new thread given the progress the beta provides. now my test sound makes a noise and my test call lanuches and stops when i tell it to, BUT! it crashes with the below. pulseaudio[3400]: resampler.c: Using float32le as working format. pulseaudio[3400]: resampler.c: Choosing speex quality setting 3. pulseaudio[3400]: source-output.c: Created output 0 "Input" on alsa_input.pci_8086_24c5_sound_card_0 with sample spec s16le 1ch 16000Hz and channel map mono pulseaudio[3400]: protocol-native.c: Final latency 199.89 ms = 99.94 ms + 99.95 ms pulseaudio[3400]: source.c: Assertion 'pa_cvolume_compatible(volume, &s->sample_spec)' failed at pulsecore/source.c:666, function pa_source_set_volume(). Aborting. any suggestions on fixing this ? tia, jackc... -- jack craig jackc(a)LinuxLightHouse.com 831-684-1375 (Office) 831-596-6924 (cell) IM: jackcraigaptos (AIM) _________________________________ This email has been ClamScanned ! www.LinuxLightHouse.com

1 0

[alsa-devel] Any patches missing for 2.6.32?
by Takashi Iwai 28 Aug '09

28 Aug '09

Hi, now 2.6.31-rc8 came up, and we are pretty close to the next merge window. If you have any patches that should go into 2.6.32, please speak up, post them now. Otherwise it'll be postponed after 2.6.32. thanks, Takashi

1 0

[alsa-devel] SNDRV_PCM_TRIGGER_STOP and audio still queued in the driver
by Jon Smirl 28 Aug '09

28 Aug '09

In the mpc5200 driver I'm receiving SNDRV_PCM_TRIGGER_STOP with audio still queued. I noticed this in an app that uses a 2MB ALSA buffer. It is filling the 2MB buffer which starts playing. After the buffer is filled I receive the STOP. This STOP comes before the hardware has a chance to play all of the music. There is still about twenty seconds of music in the driver buffer waiting to be played. How is this supposed to work? -- Jon Smirl jonsmirl(a)gmail.com

2 11

[alsa-devel] [PATCH 1/1] sound: usb-audio: support for Digidesign Mbox 2
by Damien Zammit 27 Aug '09

27 Aug '09

Provides magic boot sequence which activates the device, must loop until device is ready, (returns status bytes). Provides extra magic to set the device into 24 bit mode. -Playback works @ 24-bit -MIDI I/O works -Capture is untested, but PCM stream is active -S/PDIF (RCA) I/O is untested, but lights activate on the device S/PDIF mode works by setting the required flag in usbaudio.c. I think the device can't do both simultaneously, since different altsettings are required to set it. Remaining issues: 1) Double check endianess of 24 bit capture. 2) Check duplex stability 3) Tell ALSA to correctly detect S24_3BE on the device. Currently, speaker-test assumes S16_LE and won't allow S24_3BE mode, but jackd 0.116-2 detects and plays S24_3BE ? 4) Create a kernel module switch to select spdif/analogue I/O Playback sample format : S24_3BE MIDI format : MIDIMAN protocol Signed-off-by: Damien Zammit <damien.zammit(a)gmail.com> diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 43d7d88..3060941 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -943,7 +943,7 @@ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream { struct snd_usb_substream *subs; snd_pcm_uframes_t hwptr_done; - + subs = (struct snd_usb_substream *)substream->runtime->private_data; spin_lock(&subs->lock); hwptr_done = subs->hwptr_done; @@ -2441,6 +2441,12 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audiofor fp->altsetting, sample_width, sample_bytes); break; } + + /* Digidesign Mbox 2 workaround: + * supports S24_3BE BIG ENDIAN */ + if (chip->usb_id == USB_ID(0x0dba, 0x3000)) + pcm_format = SNDRV_PCM_FORMAT_S24_3BE; + break; case USB_AUDIO_FORMAT_PCM8: pcm_format = SNDRV_PCM_FORMAT_U8; @@ -2550,13 +2556,20 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, struct audioformat * switch (chip->usb_id) { case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ - if (device_setup[chip->index] == 0x00 && + if (device_setup[chip->index] == 0x00 && fp->altsetting == 6) pcm_format = SNDRV_PCM_FORMAT_S16_BE; else pcm_format = SNDRV_PCM_FORMAT_S16_LE; break; - default: + + /* Digidesign Mbox 2 supports SPDIF 24 bit (havent tested endianess) */ + case USB_ID(0x0dba, 0x3000): + pcm_format = SNDRV_PCM_FORMAT_S24_3BE; + break; + + default: + snd_printdd("Format III detected, setting default S16_LE\n"); pcm_format = SNDRV_PCM_FORMAT_S16_LE; } } else { @@ -2655,6 +2668,8 @@ static unsigned char parse_datainterval(struct snd_usb_audio *chip, static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip, int iface, int altno); +static int mbox2_skip_setting_quirk(struct snd_usb_audio *chip, + int iface, int altno); static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) { struct usb_device *dev; @@ -2700,13 +2715,19 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) stream = (get_endpoint(alts, 0)->bEndpointAddress & USB_DIR_IN) ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; altno = altsd->bAlternateSetting; - + /* audiophile usb: skip altsets incompatible with device_setup */ - if (chip->usb_id == USB_ID(0x0763, 0x2003) && + if (chip->usb_id == USB_ID(0x0763, 0x2003) && audiophile_skip_setting_quirk(chip, iface_no, altno)) continue; + /* Digidesign Mbox 2: skip altsets incompatible with device_setup + */ + if (chip->usb_id == USB_ID(0x0dba, 0x3000) && + mbox2_skip_setting_quirk(chip, iface_no, altno)) + continue; + /* get audio formats */ fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL); if (!fmt) { @@ -2995,7 +3016,7 @@ static int create_standard_audio_quirk(struct snd_usb_audio *chip, } /* - * Create a stream for an Edirol UA-700/UA-25/UA-4FX interface. + * Create a stream for an Edirol UA-700/UA-25/UA-4FX interface. * The only way to detect the sample rate is by looking at wMaxPacketSize. */ static int create_uaxx_quirk(struct snd_usb_audio *chip, @@ -3264,6 +3285,97 @@ static int snd_usb_extigy_boot_quirk(struct usb_device *dev, struct usb_interfac return 0; } + +#define MBOX2_FIRMWARE_SIZE 646 +#define MBOX2_BOOT_LOADING 0x01 /* Hard coded into the device */ +#define MBOX2_BOOT_READY 0x02 /* Hard coded into the device */ + +static int snd_usb_mbox2_boot_quirk(struct usb_device *dev) +{ + struct usb_host_config *config = dev->actconfig; + int err; + u8 enablemagic[3]; + u8 bootresponse; + int fwsize; + + if ((fwsize = le16_to_cpu(get_cfg_desc(config)->wTotalLength)) == MBOX2_FIRMWARE_SIZE) { + snd_printdd("Sending Digidesign Mbox 2 boot sequence...\n"); + + /* From USB Snoop, + * + * SetupPacket = RT RQ VHVL INDX SIZE + * RT = 0xRT Request Type + * RQ = 0xRQ Request + * VHVL = 0xVLVH Value in reverse byte order + * INDX = 0xDXIN Index in reverse byte order + * SIZE = 0xZESI Size in reverse byte order + * + * Magic boot code setup packet: c0 85 01 00 00 00 12 00 + * RQ RT VLVH DXIN ZESI + * becomes 0x85, 0xc0, 0x0001, 0x0000, &RETURNDATA, 0x0012, TIMEOUT + * for snd_usb_ctl_msg() + */ + +mbox2_reboot: + snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), + 0x85, 0xc0, 0x0001, 0x0000, &bootresponse, 0x0012, 1000); + + if (bootresponse == MBOX2_BOOT_LOADING) { + snd_printdd("device not ready, resending boot sequence...\n"); + goto mbox2_reboot; + } + + if (bootresponse == MBOX2_BOOT_READY) { + snd_printdd("device initialised!\n"); + + err = usb_get_descriptor(dev, USB_DT_DEVICE, 0, + &dev->descriptor, sizeof(dev->descriptor)); + config = dev->actconfig; + if (err < 0) snd_printdd("error usb_get_descriptor: %d\n", err); + err = usb_reset_configuration(dev); + if (err < 0) snd_printdd("error usb_reset_configuration: %d\n", err); + snd_printdd("mbox2_boot: new boot length = %d\n", + le16_to_cpu(get_cfg_desc(config)->wTotalLength)); + + /* Successful boot, (should reset all altsettings to 0 like windows does, but works without it) + * but now we will try sending capture/playback enable magic + * + * 80 bb 00 = 24bit mode - S24_3BE + * 44 ac 00 = 16bit mode? + * + */ + //enablemagic[0]=0x44; + //enablemagic[1]=0xac; + //enablemagic[2]=0x00; + + /* We want 24 bit mode for now */ + enablemagic[0]=0x80; + enablemagic[1]=0xbb; + enablemagic[2]=0x00; + + /* Send the magic! */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x01, 0x22, 0x0100, 0x0085, &enablemagic, 0x0003, 1000); + + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x81, 0xa2, 0x0100, 0x0085, &enablemagic, 0x0003, 1000); + + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x81, 0xa2, 0x0100, 0x0086, &enablemagic, 0x0003, 1000); + + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x81, 0xa2, 0x0100, 0x0003, &enablemagic, 0x0003, 1000); + + return 0; /* Succesful boot */ + } + + snd_printdd("unknown bootresponse, ignoring device: %d\n",bootresponse); + return -ENODEV; + } + snd_printdd("Invalid firmware size: %d\n",fwsize); + return -ENODEV; +} + static int snd_usb_audigy2nx_boot_quirk(struct usb_device *dev) { u8 buf = 1; @@ -3366,10 +3478,79 @@ static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip, if ((device_setup[chip->index] & AUDIOPHILE_SET_MASK) == AUDIOPHILE_SET_16B_48K_NOTDI && altno != 5) return 1; /* skip this altsetting */ - } + } return 0; /* keep this altsetting */ } +/* Digidesign Mbox 2 I/O selector + * Uncomment this line to enable SPDIF */ +//#define MBOX2_SPDIF_IO + +#define MBOX2_FIRMWARE_INTERFACE 0 +#define MBOX2_CONTROL_INTERFACE 1 +#define MBOX2_MIDI_INTERFACE 6 + +static int mbox2_skip_setting_quirk(struct snd_usb_audio *chip, + int iface, int altno) +{ + /* Reset ifaces 2-5 to 0 altsetting. */ + usb_set_interface(chip->dev, iface, 0); + +#ifdef MBOX2_SPDIF_IO + /* Enable SPDIF altsettings */ + switch (iface) { + case MBOX2_FIRMWARE_INTERFACE: + case MBOX2_CONTROL_INTERFACE: + return 0; /*keep*/ + break; + case 2: + if (altno == 2) return 0; else return 1; + break; + case 3: + return 1; + break; + case 4: + if (altno == 2) return 0; else return 1; + break; + case 5: + return 1; + break; + case MBOX2_MIDI_INTERFACE: + return 0; /*keep*/ + break; + default: + snd_printdd("Invalid interface for Mbox 2: %d\n",iface); + return 1; /* skip altsetting */ + } +#else + /* Use analogue I/O */ + switch (iface) { + case MBOX2_FIRMWARE_INTERFACE: + case MBOX2_CONTROL_INTERFACE: + return 0; + break; + case 2: + if (altno == 3) return 0; else return 1; + break; + case 3: + if (altno == 6) return 0; else return 1; + break; + case 4: + if (altno == 3) return 0; else return 1; + break; + case 5: + if (altno == 3) return 0; else return 1; + break; + case MBOX2_MIDI_INTERFACE: + return 0; + break; + default: + snd_printdd("Invalid interface for Mbox 2: %d\n",iface); + return 1; /* skip altsetting */ + } +#endif +} + /* * audio-interface quirks * @@ -3392,6 +3573,7 @@ static int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_MIDI_YAMAHA] = snd_usb_create_midi_interface, [QUIRK_MIDI_MIDIMAN] = snd_usb_create_midi_interface, [QUIRK_MIDI_NOVATION] = snd_usb_create_midi_interface, + [QUIRK_MIDI_MBOX2] = snd_usb_create_midi_interface, [QUIRK_MIDI_FASTLANE] = snd_usb_create_midi_interface, [QUIRK_MIDI_EMAGIC] = snd_usb_create_midi_interface, [QUIRK_MIDI_CME] = snd_usb_create_midi_interface, @@ -3425,7 +3607,7 @@ static void proc_audio_usbid_read(struct snd_info_entry *entry, struct snd_info_ { struct snd_usb_audio *chip = entry->private_data; if (!chip->shutdown) - snd_iprintf(buffer, "%04x:%04x\n", + snd_iprintf(buffer, "%04x:%04x\n", USB_ID_VENDOR(chip->usb_id), USB_ID_PRODUCT(chip->usb_id)); } @@ -3616,6 +3798,12 @@ static void *snd_usb_audio_probe(struct usb_device *dev, goto __err_val; } + /* Digidesign Mbox 2 needs magic boot sequence */ + if (id == USB_ID(0x0dba, 0x3000)) { + if (snd_usb_mbox2_boot_quirk(dev) < 0) + goto __err_val; + } + /* * found a config. now register to ALSA */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 8e7f789..21aab48 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -157,6 +157,7 @@ enum quirk_type { QUIRK_MIDI_EMAGIC, QUIRK_MIDI_CME, QUIRK_MIDI_US122L, + QUIRK_MIDI_MBOX2, QUIRK_AUDIO_STANDARD_INTERFACE, QUIRK_AUDIO_FIXED_ENDPOINT, QUIRK_AUDIO_EDIROL_UA1000, diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 0eff19c..bc11fe4 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -423,7 +423,7 @@ static void snd_usbmidi_maudio_broken_running_status_input( u8 cin = buffer[i] & 0x0f; struct usbmidi_in_port *port = &ep->ports[cable]; int length; - + length = snd_usbmidi_cin_length[cin]; if (cin == 0xf && buffer[i + 1] >= 0xf8) ; /* realtime msg: no running status change */ @@ -617,13 +617,13 @@ static struct usb_protocol_ops snd_usbmidi_standard_ops = { static struct usb_protocol_ops snd_usbmidi_midiman_ops = { .input = snd_usbmidi_midiman_input, - .output = snd_usbmidi_standard_output, + .output = snd_usbmidi_standard_output, .output_packet = snd_usbmidi_output_midiman_packet, }; static struct usb_protocol_ops snd_usbmidi_maudio_broken_running_status_ops = { .input = snd_usbmidi_maudio_broken_running_status_input, - .output = snd_usbmidi_standard_output, + .output = snd_usbmidi_standard_output, .output_packet = snd_usbmidi_output_standard_packet, }; @@ -1577,7 +1577,7 @@ static int snd_usbmidi_detect_endpoints(struct snd_usb_midi* umidi, snd_usbmidi_switch_roland_altsetting(umidi); if (endpoint[0].out_ep || endpoint[0].in_ep) - return 0; + return 0; intf = umidi->iface; if (!intf || intf->num_altsetting < 1) @@ -1615,7 +1615,7 @@ static int snd_usbmidi_detect_per_port_endpoints(struct snd_usb_midi* umidi, struct snd_usb_midi_endpoint_info* endpoints) { int err, i; - + err = snd_usbmidi_detect_endpoints(umidi, endpoints, MIDI_MAX_ENDPOINTS); for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) { if (endpoints[i].out_ep) @@ -1876,6 +1876,18 @@ int snd_usb_create_midi_interface(struct snd_usb_audio* chip, umidi->usb_protocol_ops = &snd_usbmidi_novation_ops; err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); break; + case QUIRK_MIDI_MBOX2: + /* Digidesign Mbox 2 uses MIDIMAN MIDI protocol */ + umidi->usb_protocol_ops = &snd_usbmidi_midiman_ops; + /* + * We have to make sure that the USB core looks + * again at interface 6 by calling usb_set_interface() on it. + */ + usb_set_interface(umidi->chip->dev, 6, 0); + memcpy(&endpoints[0], quirk->data, + sizeof(struct snd_usb_midi_endpoint_info)); + err = 0; + break; case QUIRK_MIDI_FASTLANE: umidi->usb_protocol_ops = &snd_usbmidi_raw_ops; /* diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index f6f201e..e961f5e 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -2082,6 +2082,60 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* DIGIDESIGN MBOX 2 */ +{ + /* Damien Zammit <damien.zammit(a)gmail.com> */ + + USB_DEVICE(0x0dba, 0x3000), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Digidesign", + .product_name = "Mbox 2", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 3, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 4, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 5, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 6, + .type = QUIRK_MIDI_MBOX2, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_ep = 0x02, + .out_cables = 0x0001, + .in_ep = 0x81, + .in_interval = 0x01, + .in_cables = 0x0001 + } + + }, + { + .ifnum = -1 + } + } + } +}, + { /* * Some USB MIDI devices don't have an audio control interface, -- 1.5.4.3

2 3

[alsa-devel] ALSA default buffer sizes (recent now being subscribed with correct address... hopefully)
by Manuel Jander 26 Aug '09

26 Aug '09

Hi there, I'm somewhat upset because audacious does not work with my AU8830. The problem is that audacious never calls any function that would give any clue about the desired buffer size. So my AU8830 ends up with 32 periods, 1 byte each. Great ! I'm not going to blame the author of that driver :P ... but: The question is, who is the entity deciding those default values ? Other soundcards seem to get reasonable defaults, but the AU8830 does not. So I would like to fix that, but I want to do that at the right place. Best Regards, and many thanks for any hint.

3 4

[alsa-devel] [PATCH] vwsnd: Fix setting of cfgval and ctlval in li_setup_dma()
by Roel Kluin 26 Aug '09

26 Aug '09

Since !LI_CCFG_* evaluates to 0, this did not change anything to cfgval and ctlval. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- // vi sound/oss/vwsnd.c +392 # define LI_CCTL_RESET 0x80000000 ... # define LI_CCTL_DMA_ENABLE 0x08000000 ... # define LI_CCFG_LOCK 0x00008000 I am not sure this is what was intended, does anyone have the specs? diff --git a/sound/oss/vwsnd.c b/sound/oss/vwsnd.c index 187f727..6713110 100644 --- a/sound/oss/vwsnd.c +++ b/sound/oss/vwsnd.c @@ -628,7 +628,7 @@ static void li_setup_dma(dma_chan_t *chan, ASSERT(!(buffer_paddr & 0xFF)); chan->baseval = (buffer_paddr >> 8) | 1 << (37 - 8); - chan->cfgval = (!LI_CCFG_LOCK | + chan->cfgval = ((chan->cfgval & ~LI_CCFG_LOCK) | SHIFT_FIELD(desc->ad1843_slot, LI_CCFG_SLOT) | desc->direction | mode | @@ -638,9 +638,9 @@ static void li_setup_dma(dma_chan_t *chan, tmask = 13 - fragshift; /* See Lithium DMA Notes above. */ ASSERT(size >= 2 && size <= 7); ASSERT(tmask >= 1 && tmask <= 7); - chan->ctlval = (!LI_CCTL_RESET | + chan->ctlval = ((chan->ctlval & ~LI_CCTL_RESET) | SHIFT_FIELD(size, LI_CCTL_SIZE) | - !LI_CCTL_DMA_ENABLE | + (chan->ctlval & ~LI_CCTL_DMA_ENABLE) | SHIFT_FIELD(tmask, LI_CCTL_TMASK) | SHIFT_FIELD(0, LI_CCTL_TPTR));

2 1

[alsa-devel] [PATCH 1/4] SOUND: atiixp, use uninitialized_var(chip)
by Jiri Slaby 26 Aug '09

26 Aug '09

To avoid wrong compiler warnings, use unitialized_var(chip) in snd_atiixp_probe's from atiixp and atiixp_modem drivers. 'chip' is unused when unset due to retval being nonzero. Signed-off-by: Jiri Slaby <jirislaby(a)gmail.com> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> --- sound/pci/atiixp.c | 2 +- sound/pci/atiixp_modem.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index d6752df..436ffea 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -1648,7 +1648,7 @@ static int __devinit snd_atiixp_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { struct snd_card *card; - struct atiixp *chip; + struct atiixp *uninitialized_var(chip); int err; err = snd_card_create(index, id, THIS_MODULE, 0, &card); diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index e7e147b..b6191f2 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -1285,7 +1285,7 @@ static int __devinit snd_atiixp_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { struct snd_card *card; - struct atiixp_modem *chip; + struct atiixp_modem *uninitialized_var(chip); int err; err = snd_card_create(index, id, THIS_MODULE, 0, &card); -- 1.6.3.3

2 4

[alsa-devel] alsa random crash
by some guy 26 Aug '09

26 Aug '09

Problem: Alsa randomly crashes after ~10 hours of use; this varies from 2 to 20 hours. Symptoms: Sound stops while media is playing (in web browser, or mplayer), or convulses sporatically. The issue is not always reproducible (i.e I can reproduce the result when I want), but it is perisistant. Info: http://www.alsa-project.org/db/?f=fe434884ea7a16b1ae7c06cfd282832971f41fd6 lspci erhttp://pastebin.ca/1542518 kernel version is gentoo-2.6.28-r5 I really appreciate it. -anonymous alsa'er. _________________________________________________________________ More storage. Better anti-spam and antivirus protection. Hotmail makes it simple. http://go.microsoft.com/?linkid=9671357

1 0

[alsa-devel] [PATCH] ASoC: SDP3430: Fix TWL GPIO6 pin mux request
by Candelaria Villareal, Jorge 25 Aug '09

25 Aug '09

Fix the write to PMBR1 register through I2C. Also, the constant which holds the value to write is now called TWL4030_GPIO6_PWM0_MUTE. This name is based on TRM to avoid confusion. Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- sound/soc/omap/sdp3430.c | 15 +++++++++++---- 1 files changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index f7e5b74..4a3f62d 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -40,8 +40,10 @@ #include "omap-pcm.h" #include "../codecs/twl4030.h" -#define TWL4030_INTBR_PMBR1 0x0D -#define EXTMUTE(value) (value << 2) +/* TWL4030 PMBR1 Register */ +#define TWL4030_INTBR_PMBR1 0x0D +/* TWL4030 PMBR1 Register GPIO6 mux bit */ +#define TWL4030_GPIO6_PWM0_MUTE(value) (value << 2) static struct snd_soc_card snd_soc_sdp3430; @@ -299,6 +301,7 @@ static struct platform_device *sdp3430_snd_device; static int __init sdp3430_soc_init(void) { int ret; + u8 pin_mux; if (!machine_is_omap_3430sdp()) { pr_debug("Not SDP3430!\n"); @@ -318,8 +321,12 @@ static int __init sdp3430_soc_init(void) *(unsigned int *)sdp3430_dai[1].cpu_dai->private_data = 2; /* McBSP3 */ /* Set TWL4030 GPIO6 as EXTMUTE signal */ - twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, EXTMUTE(0x02), - TWL4030_MODULE_INTBR); + twl4030_i2c_read_u8(TWL4030_MODULE_INTBR, &pin_mux, + TWL4030_INTBR_PMBR1); + pin_mux &= ~TWL4030_GPIO6_PWM0_MUTE(0x03); + pin_mux |= TWL4030_GPIO6_PWM0_MUTE(0x02); + twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, pin_mux, + TWL4030_INTBR_PMBR1); ret = platform_device_add(sdp3430_snd_device); if (ret) -- 1.6.0.4

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Re: [alsa-devel] Acer Aspire 7730G support
by Denis Kup 25 Aug '09

25 Aug '09

2009/8/25 Takashi Iwai <tiwai(a)suse.de> > BTW, did you check, as I suggested, whether acer-aspire-6530g with > changing the channel mode from 2ch to 6ch works for 7730g? > I guess we can merge acer-aspire-* variants to one or two models. > Can you describe in details what I need to do, please?

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Re: [alsa-devel] Acer Aspire 7730G support
by Takashi Iwai 25 Aug '09

25 Aug '09

At Mon, 24 Aug 2009 19:57:58 +0400, Denis Kup wrote: > > 2009/8/24 Takashi Iwai <tiwai(a)suse.de> > > > At Mon, 24 Aug 2009 17:10:10 +0400, > > Denis Kup wrote: > > > > > > Sorry, I was very busy last two months. We don't finish with this theme. > > > What about my patch? > > > > Please check with the latest sound GIT tree (or alsa-driver snapshot) > > whether it still work and repost the patch. There have been lots of > > changes in HD-audio driver code, so an old patch might not work any > > more. > > > > > > thanks, > > > > Takashi > > > I tested patch with alsa-driver-1.0.20.40.g13311.579.gcf341. All works as it > should. Try rather with the snapshot tarball in ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sna… The version you tried isn't the latest one. > It's diff between patched version( patch_realtek_new.c ) and unpatched: Make a patch with diff -u option (and better with -p, too). thanks, Takashi

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[alsa-devel] [PATCH] ASoC: SDP3430: Fix TWL GPIO6 pin mux request
by Candelaria Villareal, Jorge 25 Aug '09

25 Aug '09

Fix the write to PMBR1 register through I2C. Also, the constant which holds the value to write is now called TWL4030_GPIO6_PWM0_MUTE. This name is based on TRM to avoid confusion. Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- sound/soc/omap/sdp3430.c | 15 +++++++++++---- 1 files changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index f7e5b74..7ad99d2 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -40,8 +40,10 @@ #include "omap-pcm.h" #include "../codecs/twl4030.h" -#define TWL4030_INTBR_PMBR1 0x0D -#define EXTMUTE(value) (value << 2) +/* TWL4030 PMBR1 Register */ +#define TWL4030_INTBR_PMBR1 0x0D +/* TWL4030 PMBR1 Register GPIO6 mux bit */ +#define TWL4030_GPIO6_PWM0_MUTE(value) (value << 2) static struct snd_soc_card snd_soc_sdp3430; @@ -299,6 +301,7 @@ static struct platform_device *sdp3430_snd_device; static int __init sdp3430_soc_init(void) { int ret; + u8 pin_mux; if (!machine_is_omap_3430sdp()) { pr_debug("Not SDP3430!\n"); @@ -318,8 +321,12 @@ static int __init sdp3430_soc_init(void) *(unsigned int *)sdp3430_dai[1].cpu_dai->private_data = 2; /* McBSP3 */ /* Set TWL4030 GPIO6 as EXTMUTE signal */ - twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, EXTMUTE(0x02), - TWL4030_MODULE_INTBR); + twl4030_i2c_read_u8(TWL4030_MODULE_INTBR, &pin_mux, + TWL4030_INTBR_PMBR1); + pin_mux &= ~TWL4030_GPIO6_PWM0_MUTE(0x3); + pin_mux |= TWL4030_GPIO6_PWM0_MUTE(0x2); + twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, pin_mux, + TWL4030_INTBR_PMBR1); ret = platform_device_add(sdp3430_snd_device); if (ret) -- 1.6.0.4

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