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[alsa-devel] trouble using dmix with tts systems (accessibility )
by Halim Sahin 23 Mar '09

23 Mar '09

hello, I have asked this question to alsa user list but got no useful answer. I am a blind linux user and need asistive Technology to use my pc. The problem I have experienced with alsa is that it is sometimes too slow to stop previous output before starting a new one. The result of this ends in several speech outputs at a time. The problem only occurs if the dmix plugin is used. Is there a way to modify the buffer to avoid these delays??? I have tried to customize my buffer_size vallue in ~/.asoundrc but I wasn't able to to create a working setup with dmix. All help appreciated ;-) Regards halim

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[alsa-devel] [PATCH] alsa: Intel HDA output and align Ids.
by Pascal de Bruijn | Unilogic Networks B.V. 23 Mar '09

23 Mar '09

This patch does two things: Output Intel HDA Function Id in /proc/asound/cardX/codec#X Align Vendor/Subsystem/Revision Ids to 8 characters, front-padded with zeros Before: Vendor Id: 0x11d41884 Subsystem Id: 0x103c281a Revision Id: 0x100100 After: Function Id: 0x1 Vendor Id: 0x11d41884 Subsystem Id: 0x103c281a Revision Id: 0x0100100 As report on the Kernel Bugzilla #12888 Signed-off-by: Pascal de Bruijn <pascal(a)unilogicnetworks.net> diff -Nurp linux-2.6.29-rc8-orig/sound/pci/hda/hda_codec.c linux-2.6.29-rc8-func/sound/pci/hda/hda_codec.c --- linux-2.6.29-rc8-orig/sound/pci/hda/hda_codec.c 2009-03-20 16:05:15.000000000 +0100 +++ linux-2.6.29-rc8-func/sound/pci/hda/hda_codec.c 2009-03-23 10:37:30.000000000 +0100 @@ -647,9 +647,9 @@ static void /*__devinit*/ setup_fg_nodes total_nodes = snd_hda_get_sub_nodes(codec, AC_NODE_ROOT, &nid); for (i = 0; i < total_nodes; i++, nid++) { - unsigned int func; - func = snd_hda_param_read(codec, nid, AC_PAR_FUNCTION_TYPE); - switch (func & 0xff) { + codec->function_id = snd_hda_param_read(codec, nid, + AC_PAR_FUNCTION_TYPE) & 0xff; + switch (codec->function_id) { case AC_GRP_AUDIO_FUNCTION: codec->afg = nid; break; diff -Nurp linux-2.6.29-rc8-orig/sound/pci/hda/hda_codec.h linux-2.6.29-rc8-func/sound/pci/hda/hda_codec.h --- linux-2.6.29-rc8-orig/sound/pci/hda/hda_codec.h 2009-03-20 16:05:15.000000000 +0100 +++ linux-2.6.29-rc8-func/sound/pci/hda/hda_codec.h 2009-03-20 16:10:07.000000000 +0100 @@ -739,6 +739,7 @@ struct hda_codec { hda_nid_t mfg; /* MFG node id */ /* ids */ + u32 function_id; u32 vendor_id; u32 subsystem_id; u32 revision_id; diff -Nurp linux-2.6.29-rc8-orig/sound/pci/hda/hda_proc.c linux-2.6.29-rc8-func/sound/pci/hda/hda_proc.c --- linux-2.6.29-rc8-orig/sound/pci/hda/hda_proc.c 2009-03-20 16:05:15.000000000 +0100 +++ linux-2.6.29-rc8-func/sound/pci/hda/hda_proc.c 2009-03-20 16:10:07.000000000 +0100 @@ -467,8 +467,9 @@ static void print_codec_info(struct snd_ snd_iprintf(buffer, "Codec: %s\n", codec->name ? codec->name : "Not Set"); snd_iprintf(buffer, "Address: %d\n", codec->addr); - snd_iprintf(buffer, "Vendor Id: 0x%x\n", codec->vendor_id); - snd_iprintf(buffer, "Subsystem Id: 0x%x\n", codec->subsystem_id); + snd_iprintf(buffer, "Function Id: 0x%x\n", codec->function_id); + snd_iprintf(buffer, "Vendor Id: 0x%08x\n", codec->vendor_id); + snd_iprintf(buffer, "Subsystem Id: 0x%08x\n", codec->subsystem_id); snd_iprintf(buffer, "Revision Id: 0x%x\n", codec->revision_id); if (codec->mfg)

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[alsa-devel] How to find sampling rate supported
by reetesh chandran 23 Mar '09

23 Mar '09

Hi all, I am new to the group and to Alsa. For our project, we have to find the max sampling rate and other pcm hardware parameters that the connected USB headset supports. I looked at the alsa library, but could not find an API which returned this information. I found snd_pcm_hw_params_test_* function to test a specific value is supported, but I was wondering if there was a way to query the device to get the values. Thanks in advance, Reetesh

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Re: [alsa-devel] Bugs on aspire one A150
by Andreas Mohr 23 Mar '09

23 Mar '09

Hi, On Mon, Mar 16, 2009 at 05:19:38PM +0100, Takashi Iwai wrote: > At Mon, 16 Mar 2009 17:06:35 +0100, > 私 wrote: > > > > > > What are "sliders"? > > > > > > Umm, volume level controls. > > > > Yes but there are many of such :) > > > > More exactly, from the driver perspective, there are no volume > > controls but only there are control elements with integer values. > > Do you mean "Capture Volume" control or which one? Hmm, ok, this needs to be more precise: In gamix (codec "HDA Intel : Realtek ALC268"), the Capture Volume control. > > And, is the behavior consistent regardless of the value high, i.e. > > the key is only whether the values for both channels are identical? > > BTW, what if you record with the following definition? > Put the below to ~/.asoundrc > > pcm.imix { > type plug > slave.pcm "hw" > ttable.0.0 0.5 > ttable.0.1 -0.5 > } > > and record like > > % aplay -Dimix -c1 foo.wav Does NOT exhibit the "equal sliders == no sound" bug (apart from this sliders are acting normally, i.e. slider low == no sound), despite being a "plug" type definition (this is what you wanted to discern, right? ;). Thanks! Andreas

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[alsa-devel] MSI GX620 support
by okias 22 Mar '09

22 Mar '09

MSI GX620X-061CZ, but probably is this audio chip with this configuration on all GX620 Connectors: Green, standart jack 3.5 headphones Blue, line-in Pink, mic Grey, side DOLBY HOME THEATER KO 127 3stack-dig 3-jack with SPDIF I/O *OK 128 6stack-dig 6-jack digital with SPDIF I/O // work SIDE+jack, but SIDE can't be configured, not deactivating integrated repro *OK 129 3stack-6ch 3-jack 6-channel // headphone power off integrated repro, 6ch working, no 8ch OK 130 3stack-6ch-dig 3-jack 6-channel with SPDIF I/O //same 3stacj-6ch KO 131 6stack-dig-demo 6-jack digital for Intel demo board OK 132 acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc)//jack+side, but SIDE can't be configured, headphone power off integr., notworking 6ch KO 133 acer-aspire Acer Aspire 9810 *OK 134 acer-aspire-4930g Acer Aspire 4930G // Front switch shutting down both repro (external in jack, integrated), 6channels + SIDE (should be 8ch), and if is plugged something into jack integrated repro still playing KO 135 medion Medion Laptops KO 136 medion-md2 Medion MD2 !! 137 targa-dig Targa/MSI // If is plugged jack, then integrated repro power down! but missing settings to keep integrated repro playing and 8ch isn't present(side not working). KO 138 targa-2ch-dig Targs/MSI with 2-channel KO 139 laptop-eapd 3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE) // SIDE working, but 6ch conf not, only side + repro output KO 140 lenovo-101e Lenovo 101E // nothing new... less settings, no side... no 6ch KO 141 lenovo-nb0763 Lenovo NB0763 KO 142 lenovo-ms7195-dig Lenovo MS7195 KO 143 lenovo-sky Lenovo Sky KO 144 haier-w66 Haier W66 KO 145 3stack-hp HP machines with 3stack (Lucknow, Samba boards) // 4. channel not working, etc. KO 146 6stack-dell Dell machines with 6stack (Inspiron 530) // if is activated 8ch conf, side is activated, but 4 and 6th channel not working with this conf KO 147 mitac Mitac 8252D KO 148 clevo-m720 Clevo M720 laptop series KO 149 fujitsu-pi2515 Fujitsu AMILO Pi2515 *OK 150 fujitsu-xa3530 Fujitsu AMILO XA3530 // 6ch mode working, but SIDE working even if is set only 6ch KO 151 3stack-6ch-intel Intel DG33* boards OK 152 auto auto-config reading BIOS (default) // SIDE+JACK, BOTH MIC OK, can't be configured to 4,6 or 8ch, if is pluged into JACK repro then integrated playing... OK means, it's in usable state, KO mean something important doesn't work, or nothing new in compare to before tested configurations. *OK means very good configurations 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) Subsystem: Micro-Star International Co., Ltd. Device 6510 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 32 bytes Interrupt: pin A routed to IRQ 22 Region 0: Memory at f8ef8000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] MSI: Mask- 64bit+ Count=1/1 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <64ns, L1 <1us ExtTag- RBE- FLReset+ DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #0, Speed unknown, Width x0, ASPM unknown, Latency L0 <64ns, L1 <1us ClockPM- Surprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; Disabled- Retrain- CommClk- ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed unknown, Width x0, TrErr- Train- SlotClk- DLActive- BWMgmt- ABWMgmt- Capabilities: [100] Virtual Channel <?> Capabilities: [130] Root Complex Link <?> Kernel driver in use: HDA Intel If you need any addional information, testing patchs (gentoo), please contact me :-) okias -- Jabber/XMPP: okias(a)isgeek.info Webové stránky: http://www.jabber.xf.cz

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[alsa-devel] [PATCH] alsa: Intel HDA output and align Ids.
by Pascal de Bruijn | Unilogic Networks B.V. 20 Mar '09

20 Mar '09

Output Intel HDA Function Id in /proc/asound/cardX/codec#X Align Vendor/Subsystem/Revision Ids to 8 characters, front-padded with zeros As submitted to Kernel Bugzilla #12888 Signed-off-by: Pascal de Bruijn <pascal(a)unilogicnetworks.net> diff -Nurp linux-2.6.28.8-orig/sound/pci/hda/hda_codec.c linux-2.6.28.8-func/sound/pci/hda/hda_codec.c --- linux-2.6.28.8-orig/sound/pci/hda/hda_codec.c 2009-03-17 01:50:03.000000000 +0100 +++ linux-2.6.28.8-func/sound/pci/hda/hda_codec.c 2009-03-17 10:39:45.000000000 +0100 @@ -519,9 +519,8 @@ static void __devinit setup_fg_nodes(str total_nodes = snd_hda_get_sub_nodes(codec, AC_NODE_ROOT, &nid); for (i = 0; i < total_nodes; i++, nid++) { - unsigned int func; - func = snd_hda_param_read(codec, nid, AC_PAR_FUNCTION_TYPE); - switch (func & 0xff) { + codec->function_id = snd_hda_param_read(codec, nid, AC_PAR_FUNCTION_TYPE) & 0xff; + switch (codec->function_id) { case AC_GRP_AUDIO_FUNCTION: codec->afg = nid; break; diff -Nurp linux-2.6.28.8-orig/sound/pci/hda/hda_codec.h linux-2.6.28.8-func/sound/pci/hda/hda_codec.h --- linux-2.6.28.8-orig/sound/pci/hda/hda_codec.h 2009-03-17 01:50:03.000000000 +0100 +++ linux-2.6.28.8-func/sound/pci/hda/hda_codec.h 2009-03-17 10:39:45.000000000 +0100 @@ -693,6 +693,7 @@ struct hda_codec { hda_nid_t mfg; /* MFG node id */ /* ids */ + u32 function_id; u32 vendor_id; u32 subsystem_id; u32 revision_id; diff -Nurp linux-2.6.28.8-orig/sound/pci/hda/hda_proc.c linux-2.6.28.8-func/sound/pci/hda/hda_proc.c --- linux-2.6.28.8-orig/sound/pci/hda/hda_proc.c 2009-03-17 01:50:03.000000000 +0100 +++ linux-2.6.28.8-func/sound/pci/hda/hda_proc.c 2009-03-17 10:40:08.000000000 +0100 @@ -520,9 +520,10 @@ static void print_codec_info(struct snd_ snd_hda_get_codec_name(codec, buf, sizeof(buf)); snd_iprintf(buffer, "Codec: %s\n", buf); snd_iprintf(buffer, "Address: %d\n", codec->addr); - snd_iprintf(buffer, "Vendor Id: 0x%x\n", codec->vendor_id); - snd_iprintf(buffer, "Subsystem Id: 0x%x\n", codec->subsystem_id); - snd_iprintf(buffer, "Revision Id: 0x%x\n", codec->revision_id); + snd_iprintf(buffer, "Function Id: 0x%x\n", codec->function_id); + snd_iprintf(buffer, "Vendor Id: 0x%08x\n", codec->vendor_id); + snd_iprintf(buffer, "Subsystem Id: 0x%08x\n", codec->subsystem_id); + snd_iprintf(buffer, "Revision Id: 0x%08x\n", codec->revision_id); if (codec->mfg) snd_iprintf(buffer, "Modem Function Group: 0x%x\n", codec->mfg);

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Re: [alsa-devel] [PATCH 6/9] ARM: DaVinci: ASoC: Adds McASP clock support for TI DM646X processor
by Medisetty, Naresh 19 Mar '09

19 Mar '09

> On Mon, Mar 16, 2009 at 08:05:25AM -0400, Naresh Medisetty wrote: > > > + CLK("soc-audio.0", NULL, &mcasp0_clk), > > + CLK("soc-audio.1", NULL, &mcasp1_clk), > > Assuming this is a shorthand for defining two platform devices for two > ASoC cards I'd suggest that you instead move towards registering clocks > for the DAIs and probing them as platform devices - take a look at how > the PXA AC97 driver now does this for an example. The soc-audio device > will hopefully get removed at some point and you can't guarantee that > the user won't want to put the two DAIs into a single sound card. Yes, This is a shorthand for defining two platform devices for two ASoC cards. I will re-work on this and submit a patch

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[alsa-devel] [PATCH 1/9] ARM: DaVinci: ASoC: Adds ASoC driver support for TI DM646X EVM platform
by Naresh Medisetty 19 Mar '09

19 Mar '09

Adds ASoC driver support for TI DM646X EVM platform Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git;a=commi…. sound/soc/davinci/davinci-evm.c | 131 ++++++++++++++++++++++++++++++++++++--- 1 files changed, 121 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 95c44bf..9e832c9 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -25,10 +25,15 @@ #include <mach/asp.h> #include <mach/edma.h> #include <mach/mux.h> +#include <mach/cpu.h> + #include "../codecs/tlv320aic3x.h" +#include "../codecs/codec_stubs.h" #include "davinci-pcm.h" #include "davinci-i2s.h" +#include "davinci-i2s-mcasp.h" + #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ @@ -43,14 +48,14 @@ 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()) + if (cpu_is_davinci_dm355() || cpu_is_davinci_dm646x()) sysclk = 27000000; /* ASP0 in DM6446 EVM is clocked by U55, as configured by * board-dm644x-evm.c using GPIOs from U18. There are six * options; here we "know" we use a 48 KHz sample rate. */ - else if (machine_is_davinci_evm()) + else if (cpu_is_davinci_dm644x()) sysclk = 12288000; else @@ -144,6 +149,25 @@ static struct snd_soc_dai_link evm_dai = { .ops = &evm_ops, }; +static struct snd_soc_dai_link dm6467_evm_dai[] = { + { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai = davinci_iis_mcasp_dai, + .codec_dai = &aic3x_dai, + .init = evm_aic3x_init, + .ops = &evm_ops, + }, + { + .name = "MCASP SPDIF", + .stream_name = "spdif", + .cpu_dai = &davinci_iis_mcasp_dai[1], + .codec_dai = dit_stub_dai, + .ops = &evm_ops, + }, +}; + + /* davinci-evm audio machine driver */ static struct snd_soc_card snd_soc_card_evm = { .name = "DaVinci EVM", @@ -152,12 +176,27 @@ static struct snd_soc_card snd_soc_card_evm = { .num_links = 1, }; +/* davinci dm6467 evm audio machine driver */ +static struct snd_soc_card dm6467_snd_soc_card_evm = { + .name = "DaVinci DM6467 EVM", + .platform = &davinci_soc_platform, + .dai_link = dm6467_evm_dai, + .num_links = 2, +}; + /* evm audio private data */ static struct aic3x_setup_data evm_aic3x_setup = { .i2c_bus = 1, .i2c_address = 0x1b, }; +/* dm6467 evm audio private data */ +static struct aic3x_setup_data dm6467_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, @@ -165,6 +204,14 @@ static struct snd_soc_device evm_snd_devdata = { .codec_data = &evm_aic3x_setup, }; +/* 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, +}; + + /* DM6446 EVM uses ASP0; line-out is a pair of RCA jacks */ static struct resource evm_snd_resources[] = { { @@ -174,11 +221,64 @@ static struct resource evm_snd_resources[] = { }, }; +static struct resource dm6467_evm_snd_resources[] = { + { + .start = DAVINCI_DM646X_MCASP0_REG_BASE, + .end = DAVINCI_DM646X_MCASP0_REG_BASE + (SZ_1K << 1) - 1, + .flags = IORESOURCE_MEM, + }, + { + .start = DAVINCI_DM646X_MCASP1_REG_BASE, + .end = DAVINCI_DM646X_MCASP1_REG_BASE + (SZ_1K << 1) - 1, + .flags = IORESOURCE_MEM, + }, +}; + + +static u8 dm6467_iis_serializer_direction[] = { + TX_MODE, RX_MODE, INACTIVE_MODE, INACTIVE_MODE, +}; + +static u8 dm6467_dit_serializer_direction[] = { + TX_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, +}; + + static struct evm_snd_platform_data evm_snd_data = { .tx_dma_ch = DAVINCI_DMA_ASP0_TX, .rx_dma_ch = DAVINCI_DMA_ASP0_RX, }; +static struct evm_snd_platform_data dm6467_evm_snd_data[] = { + { + .clk_name = "McASPCLK0", + .tx_dma_ch = DAVINCI_DM646X_DMA_MCASP0_AXEVT0, + .rx_dma_ch = DAVINCI_DM646X_DMA_MCASP0_AREVT0, + .tx_dma_offset = 0x400, + .rx_dma_offset = 0x400, + .op_mode = DAVINCI_MCASP_IIS_MODE, + .num_serializer = 4, + .tdm_slots = 2, + .serial_dir = dm6467_iis_serializer_direction, + .eventq_no = EVENTQ_0, + .codec_fmt = SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF, + }, + { + .clk_name = "McASPCLK1", + .tx_dma_ch = DAVINCI_DM646X_DMA_MCASP1_AXEVT1, + .rx_dma_ch = -1, + .tx_dma_offset = 0x400, + .rx_dma_offset = 0, + .op_mode = DAVINCI_MCASP_DIT_MODE, + .num_serializer = 4, + .tdm_slots = 32, + .serial_dir = dm6467_dit_serializer_direction, + .eventq_no = EVENTQ_0, + .codec_fmt = SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_IB_NF, + }, +}; + + /* DM335 EVM uses ASP1; line-out is a stereo mini-jack */ static struct resource dm335evm_snd_resources[] = { { @@ -199,23 +299,33 @@ static int __init evm_init(void) { struct resource *resources; struct evm_snd_platform_data *data; + struct snd_soc_device *evm_snd_dev_data; int index; int ret; + int res_size; - if (machine_is_davinci_evm()) { + if (cpu_is_davinci_dm644x()) { davinci_cfg_reg(DM644X_MCBSP); - + evm_snd_dev_data = &evm_snd_devdata; resources = evm_snd_resources; data = &evm_snd_data; + res_size = ARRAY_SIZE(evm_snd_resources); index = 0; - } else if (machine_is_davinci_dm355_evm()) { + } else if (cpu_is_davinci_dm355()) { /* we don't use ASP1 IRQs, or we'd need to mux them ... */ davinci_cfg_reg(DM355_EVT8_ASP1_TX); davinci_cfg_reg(DM355_EVT9_ASP1_RX); - + evm_snd_dev_data = &evm_snd_devdata; resources = dm335evm_snd_resources; + res_size = ARRAY_SIZE(dm335evm_snd_resources); data = &dm335evm_snd_data; index = 1; + } else if (cpu_is_davinci_dm646x()) { + evm_snd_dev_data = &dm6467_evm_snd_devdata; + resources = dm6467_evm_snd_resources; + res_size = ARRAY_SIZE(dm6467_evm_snd_resources); + data = dm6467_evm_snd_data; + index = 0; } else return -EINVAL; @@ -223,11 +333,12 @@ static int __init evm_init(void) if (!evm_snd_device) return -ENOMEM; - platform_set_drvdata(evm_snd_device, &evm_snd_devdata); - evm_snd_devdata.dev = &evm_snd_device->dev; - platform_device_add_data(evm_snd_device, data, sizeof(data)); + platform_set_drvdata(evm_snd_device, evm_snd_dev_data); + evm_snd_dev_data->dev = &evm_snd_device->dev; + evm_snd_device->dev.platform_data = data; - ret = platform_device_add_resources(evm_snd_device, resources, 1); + ret = platform_device_add_resources(evm_snd_device, + resources, res_size); if (ret) { platform_device_put(evm_snd_device); return ret; -- 1.5.6

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

19 Mar '09

The following changes since commit da88b48b84e1a504b6a19aff9d5b8236a59e228a: Mark Brown (1): Merge branch 'pxa-ssp' into for-2.6.30 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Jarkko Nikula (2): ASoC: OMAP: N810: Mark not connected input pins ASoC: OMAP: N810: Add more jack functions Lopez Cruz, Misael (1): ASoC: Declare Headset as Mic and Headphone widgets for SDP3430 Mark Brown (3): ASoC: Further optimise WM8400 bias configuration sequence ASoC: Add separate AVDD for WM8400 ASoC: Add FLL support for WM8400 sound/soc/codecs/wm8400.c | 159 ++++++++++++++++++++++++++++++++++++++------- sound/soc/omap/n810.c | 33 +++++++++- sound/soc/omap/sdp3430.c | 24 +++++-- 3 files changed, 181 insertions(+), 35 deletions(-)

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[alsa-devel] [PATCH 3/3] ASoC: pxa-ssp: Don't use SSCR0_SerClkDiv and SSCR0_SCR
by Philipp Zabel 19 Mar '09

19 Mar '09

Those macros are just screwed as soon as CONFIG_PXA25x is enabled. Signed-off-by: Philipp Zabel <philipp.zabel(a)gmail.com> --- sound/soc/pxa/pxa-ssp.c | 30 ++++++++++++++++++++++++++---- 1 files changed, 26 insertions(+), 4 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index c7c1996..cc8dbd4 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -283,9 +283,32 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) static void ssp_set_scr(struct ssp_dev *dev, u32 div) { struct ssp_device *ssp = dev->ssp; - u32 sscr0 = ssp_read_reg(dev->ssp, SSCR0) & ~SSCR0_SCR; + u32 sscr0 = ssp_read_reg(ssp, SSCR0); + + if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) { + sscr0 &= ~0x0000ff00; + sscr0 |= ((div - 2)/2) << 8; /* 2..512 */ + } else { + sscr0 &= ~0x000fff00; + sscr0 |= (div - 1) << 8; /* 1..4096 */ + } + ssp_write_reg(ssp, SSCR0, sscr0); +} - ssp_write_reg(ssp, SSCR0, (sscr0 | SSCR0_SerClkDiv(div))); +/** + * ssp_get_clkdiv - get SSP clock divider + */ +static u32 ssp_get_scr(struct ssp_dev *dev) +{ + struct ssp_device *ssp = dev->ssp; + u32 sscr0 = ssp_read_reg(ssp, SSCR0); + u32 div; + + if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) + div = ((sscr0 >> 8) & 0xff) * 2 + 2; + else + div = ((sscr0 >> 8) & 0xfff) + 1; + return div; } /* @@ -674,8 +697,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, case SND_SOC_DAIFMT_I2S: sspsp = ssp_read_reg(ssp, SSPSP); - if (((sscr0 & SSCR0_SCR) == SSCR0_SerClkDiv(4)) && - (width == 16)) { + if ((ssp_get_scr(ssp) == 4) && (width == 16)) { /* This is a special case where the bitclk is 64fs * and we're not dealing with 2*32 bits of audio * samples. -- 1.6.2

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