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[alsa-devel] [PATCH v4 0/4] ARM: DaVinci: ASoC: Audio support for DM646x
by Chaithrika U S 07 Jun '09

07 Jun '09

From: Chaithrika U S <chaithrika(a)ti.com> Add audio support for DM646x Signed-off-by: Naresh Medisetty <naresh(a)ti.com> Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- This patch set adds audio support for DM646x This updated version has included most of the review comments received for the previous version of the patch set. Patch 1: Registers the platform devices for ASP on dm355, dm644x and dm646x This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git Patch 2: Introduces platform driver model for dm644x and dm355 Applies against ALSA mainline - to the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git Patch 3: Adds driver support for the two instances of McASP on TI's DM646x Applies against ALSA mainline - to the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git Patch 4: Add machine driver support for DM646x Applies against ALSA mainline - to the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git These patches are based on the work done by Steve Chen <schen(a)mvista.com> and Pavel Kiryukhin <pkiryukhin(a)ru.mvista.com> on Monstavista 2.6.18 kernel.

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[alsa-devel] Testers wanted: New SB X-Fi driver
by Gernot Kohlhaas 07 Jun '09

07 Jun '09

Hi! I found a little strange thing: Recording from microphone is only possible, when you unmute the "Line-In" playback-device (and set microphone as capture device, of course). Oh, and by the way: Is there any possibility to apply a low pass filter to the subwoofer-channel without using ladspa (xmms and audacious won't play songs with my low pass filter. Mplayer, xine and aplay are working fine). Thanks again for this driver :) -- http://www.gnupp.de/verschluesselung/warum.html Key ID: 0x4C0B2250 Key: http://home.arcor.de/mr_maniac/ANDERES/public.gpg

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[alsa-devel] [RFC/PATCH] asoc: pxa-ssp.c -> enchance the I2S enchancement :)
by Daniel Ribeiro 06 Jun '09

06 Jun '09

Hi, this patch is what i need to use Paul's pxa-ssp.c I2S enchancements on my board. Basically, it doesn't enforce the "fake" network mode, as i dont need it. And i need _real_ network mode, so i prefer to set SSCR0_MOD only when really using network mode. And it uses PXA_SSP_FRM_??FS to actually set SSCR0_DataSize() too. This allows me to use 32bit frames for 16 bit stereo audio. (No, I don't want to fake network mode for this, and SSCR0_MOD doesn't work as expected when PXA is slave). This patch is incremental to the recent patch from Paul Shen. Signed-off-by: Daniel Ribeiro <drwyrm(a)gmail.com> diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index ed0906e..762cb93 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -579,7 +579,7 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: case SND_SOC_DAIFMT_LEFT_J: - sscr0 |= SSCR0_PSP | SSCR0_MOD; + sscr0 |= SSCR0_PSP; sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; /* See hw_params() */ @@ -655,7 +655,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, int dma = 0, chn = params_channels(params); u32 sscr0; u32 sspsp; - int frame_width; + int frame_width = 0; int width = snd_pcm_format_physical_width(params_format(params)); int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; @@ -698,7 +698,6 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); break; } - ssp_write_reg(ssp, SSCR0, sscr0); switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: @@ -742,6 +741,15 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, break; } + if (frame_width) { + sscr0 &= ~(SSCR0_EDSS | SSCR0_DataSize(16)); + if (frame_width % 48 == 0) + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(8); + else + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(16); + } + ssp_write_reg(ssp, SSCR0, sscr0); + /* When we use a network mode, we always require TDM slots * - complain loudly and fail if they've not been set up yet. */ -- Daniel Ribeiro

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[alsa-devel] Testers wanted: New SB X-Fi driver
by MonsieurWeller 06 Jun '09

06 Jun '09

Hello. Works fine, but the external rack of the "x-fi fatality" didn't seems to work. I plug my headphone on it and i have no sound. Thanks.

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[alsa-devel] [PATCH] ASoc: codec tlv320aic23 fix bogus divide by 0 message
by Troy Kisky 06 Jun '09

06 Jun '09

Some code analyzer software mistakenly gives divide by 0 error messages for these lines. This patch will end its confusion. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> --- sound/soc/codecs/tlv320aic23.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 9fcbb9c..0b8dcb5 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -273,14 +273,14 @@ static const unsigned short sr_valid_mask[] = { * Every divisor is a factor of 11*12 */ #define SR_MULT (11*12) -#define A(x) (x) ? (SR_MULT/x) : 0 +#define A(x) (SR_MULT/x) static const unsigned char sr_adc_mult_table[] = { - A(2), A(2), A(12), A(12), A(0), A(0), A(3), A(1), - A(2), A(2), A(11), A(11), A(0), A(0), A(0), A(1) + A(2), A(2), A(12), A(12), 0, 0, A(3), A(1), + A(2), A(2), A(11), A(11), 0, 0, 0, A(1) }; static const unsigned char sr_dac_mult_table[] = { - A(2), A(12), A(2), A(12), A(0), A(0), A(3), A(1), - A(2), A(11), A(2), A(11), A(0), A(0), A(0), A(1) + A(2), A(12), A(2), A(12), 0, 0, A(3), A(1), + A(2), A(11), A(2), A(11), 0, 0, 0, A(1) }; static unsigned get_score(int adc, int adc_l, int adc_h, int need_adc, -- 1.5.4.3

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Re: [alsa-devel] [PATCH 1/4] ASoC: pxa-ssp: enhance I2S and add Left_J support
by Paul Shen 06 Jun '09

06 Jun '09

Hi Daniel, This is Paul Shen , I will substitute boshen9(a)gmail.com for bshen9(a)marvell.com to answer open source mails. > From: Daniel Mack <daniel(a)caiaq.de> > Date: Wed, Jun 3, 2009 at 9:18 PM > Subject: Re: [alsa-devel] [PATCH 1/4] ASoC: pxa-ssp: enhance I2S and add Left_J support > To: Eric Miao <eric.y.miao(a)gmail.com> > Cc: alsa-devel(a)alsa-project.org, linux-arm-kernel <linux-arm-kernel(a)lists.arm.linux.org.uk>, Mark Brown <broonie(a)sirena.org.uk> > > > On Wed, Jun 03, 2009 at 08:33:42PM +0800, Eric Miao wrote: > > Make the pxa I2S configuration generic, add support for Left_J, add > > support for variable frame width like 32fs, 48fs, 64fs and 96fs > > > > Signed-off-by: Paul Shen <bshen9(a)marvell.com> > > Signed-off-by: Eric Miao <eric.miao(a)marvell.com> > > Cc: Daniel Mack <daniel(a)caiaq.de> > > --- > > arch/arm/mach-pxa/include/mach/regs-ssp.h | 14 +++--- > > sound/soc/pxa/pxa-ssp.c | 62 ++++++++++++++-------------- > > sound/soc/pxa/pxa-ssp.h | 9 ++++ > > 3 files changed, 47 insertions(+), 38 deletions(-) > > Ok, I tried that code on my board and this is what I had to change > there: > > The tdm time slot configuration needs to be set again in my board support > code just like in your example: snd_soc_set_tdm_slot(cpu_dai, 3, 2). And > the PXA_SSP_DIV_SCR value needed to be doubled from 4 to 8. > I tested with below codes to set the cpu_dai : format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS | PXA_SSP_FRM_64FS; snd_soc_dai_set_fmt(cpu_dai, format); snd_soc_dai_set_tdm_slot(cpu_dai, 3, 2); snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 0); snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); It is no need to "snd_soc_dai_set_pll(cpu_dai, 0, 0, clk)" , for your case ,no ssp-on-chip pll configuration is needed. On my littleton platform,it is ok, and I did not modify the PXA_SSP_DIV_SCR from 4 to 8. I dumped the ssp register: SSCR0 0xa10003ff SSCR1 0x00e01d80 SSTO 0x00000000 SSPSP 0x31a00084 SSSR 0x0000f0fc SSACD 0x00000000 SSACDD 0x00000000 How about your registers ? > With that changes, LRCLK is 44100Hz when configured to 44100Hz. But the > bitclk is not 64fs anymore but 32fs only (1.41Mhz). Is there any > implementation details I miss? What does your codec clock config look > like? > Do you mean it only down to 32fs when LRCLK is 44100HZ ? And when LRCLK is 48000HZ, all is OK ? > Another small thing: > > CC sound/soc/pxa/pxa-ssp.o > sound/soc/pxa/pxa-ssp.c:186: warning: 'ssp_get_scr' defined but not used > > Daniel > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > >

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[alsa-devel] [PATCH] Palm interrupt driven touchscreen driver
by Marek Vasut 05 Jun '09

05 Jun '09

Hi, this patch adds interrupt driven touchscreen driver. It's an addition to mainstone-wm97xx, but does some reorganization of the code there. Mark, it's the same patch we discussed on IRC, so it should be OK. Please consider applying, it's against Eric's tree. Thanks

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[alsa-devel] [PATCH v4 4/4] ARM: DaVinci: ASoC: Add machine driver support for DM646x
by Chaithrika U S 05 Jun '09

05 Jun '09

This patch does the following: (1) Add support for the DM646x machine (2) Modifications required to introduce the platform driver model to get platform data for all the machines including dm355 and dm644x. Signed-off-by: Steve Chen <schen(a)mvista.com> Signed-off-by: Pavel Kiryukhin <pkiryukhin(a)ru.mvista.com> Signed-off-by: Naresh Medisetty <naresh(a)ti.com> Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- This patch applies against ALSA mainline - the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git sound/soc/davinci/davinci-evm.c | 102 ++++++++++++++++++-------------------- 1 files changed, 48 insertions(+), 54 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 58fd1cb..f3bb6f6 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -27,9 +27,10 @@ #include <mach/mux.h> #include "../codecs/tlv320aic3x.h" +#include "../codecs/spdif_transciever.h" #include "davinci-pcm.h" #include "davinci-i2s.h" - +#include "davinci-mcasp.h" #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF) @@ -43,7 +44,7 @@ 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 (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm()) sysclk = 27000000; /* ASP0 in DM6446 EVM is clocked by U55, as configured by @@ -144,6 +145,24 @@ 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_mcasp_dai[DAVINCI_MCASP_I2S_DAI], + .codec_dai = &aic3x_dai, + .init = evm_aic3x_init, + .ops = &evm_ops, + }, + { + .name = "McASP", + .stream_name = "spdif", + .cpu_dai = &davinci_mcasp_dai[DAVINCI_MCASP_DIT_DAI], + .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 +171,26 @@ 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 = ARRAY_SIZE(dm6467_evm_dai), +}; + /* 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, @@ -165,60 +198,30 @@ static struct snd_soc_device evm_snd_devdata = { .codec_data = &evm_aic3x_setup, }; -/* DM6446 EVM uses ASP0; line-out is a pair of RCA jacks */ -static struct resource evm_snd_resources[] = { - { - .start = DAVINCI_ASP0_BASE, - .end = DAVINCI_ASP0_BASE + SZ_8K - 1, - .flags = IORESOURCE_MEM, - }, -}; - -static struct evm_snd_platform_data evm_snd_data = { - .tx_dma_ch = DAVINCI_DMA_ASP0_TX, - .rx_dma_ch = DAVINCI_DMA_ASP0_RX, -}; - -/* DM335 EVM uses ASP1; line-out is a stereo mini-jack */ -static struct resource dm335evm_snd_resources[] = { - { - .start = DAVINCI_ASP1_BASE, - .end = DAVINCI_ASP1_BASE + SZ_8K - 1, - .flags = IORESOURCE_MEM, - }, -}; - -static struct evm_snd_platform_data dm335evm_snd_data = { - .tx_dma_ch = DAVINCI_DMA_ASP1_TX, - .rx_dma_ch = DAVINCI_DMA_ASP1_RX, +/* 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, }; static struct platform_device *evm_snd_device; static int __init evm_init(void) { - struct resource *resources; - unsigned num_resources; - struct evm_snd_platform_data *data; + struct snd_soc_device *evm_snd_dev_data; int index; int ret; if (machine_is_davinci_evm()) { - davinci_cfg_reg(DM644X_MCBSP); - - resources = evm_snd_resources; - num_resources = ARRAY_SIZE(evm_snd_resources); - data = &evm_snd_data; + evm_snd_dev_data = &evm_snd_devdata; index = 0; } else if (machine_is_davinci_dm355_evm()) { - /* 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); - - resources = dm335evm_snd_resources; - num_resources = ARRAY_SIZE(dm335evm_snd_resources); - data = &dm335evm_snd_data; + evm_snd_dev_data = &evm_snd_devdata; index = 1; + } else if (machine_is_davinci_dm6467_evm()) { + evm_snd_dev_data = &dm6467_evm_snd_devdata; + index = 0; } else return -EINVAL; @@ -226,17 +229,8 @@ 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)); - - ret = platform_device_add_resources(evm_snd_device, resources, - num_resources); - if (ret) { - platform_device_put(evm_snd_device); - return ret; - } - + platform_set_drvdata(evm_snd_device, evm_snd_dev_data); + evm_snd_dev_data->dev = &evm_snd_device->dev; ret = platform_device_add(evm_snd_device); if (ret) platform_device_put(evm_snd_device); -- 1.5.6

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[alsa-devel] M-Audio Transit: Running asynchronous USB?
by Klaus Schulz 05 Jun '09

05 Jun '09

Hi there. I would like to get a confirmation on below. One of my USB soundcards is a M-Audio Transit. cat /proc/asound/card0/stream0 lists below: M-Audio Transit USB at usb-0000:00:1d.2-1.3, full speed : USB Audio Playback: Status: Running Interface = 1 Altset = 3 URBs = 2 [ 1 1 ] Packet Size = 196 Momentary freq = 44100 Hz (0x2c.199a) Interface 1 Altset 1 Format: 0x20 (24 bits) Channels: 2 Endpoint: 3 OUT (ADAPTIVE) Rates: 48001 - 96000 (continuous) Interface 1 Altset 2 Format: 0x20 (24 bits) Channels: 2 Endpoint: 3 OUT (NONE) Rates: 8000 - 48000 (continuous) Interface 1 Altset 3 Format: 0x2 (16 bits) Channels: 2 Endpoint: 3 OUT (ASYNC) Rates: 8000 - 48000 (continuous) Do I interprete it right that the Transit runs in asynchronous USB mode up to 48khz and 16bit? Does above also mean that <48/24 is not supported? THX Klaus

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[alsa-devel] How to send control codes to a USB device?
by Klaus Schulz 05 Jun '09

05 Jun '09

Hi there. Is there a way to sent control codes to a USB device, without using a specific driver . I would like to try to set latency and output modes of my M-Audio Transit by sending a control code as additional part of the UDEV madfuload firmware initialzation. How can I do something like that? If this works in general I just need to figure out the required control codes. THX for your support Klaus

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