June 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Writing a driver for the Blackfin BF537 with two CS42448 on SPORT1
by Adam Rosenberg 25 Jun '10

25 Jun '10

I am writing my first ALSA driver for uClinux on Blackfin BF537. I have a custom board with two Cirrus Logic CS42448 chips connected to the primary and secondary data lines of SPORT1. I was basing my driver off of the driver for the Analog Devices AD1836 but it does not look like that driver was written using the latest practices. I already have the code modified well enough to communicate with both chips and produce some noise, but it is obvious that I am not filling the DMA buffer properly. Please let me know where I can find a driver that would be a better example for me to reference. When I have completed the driver I will also need help on where to submit it so that it will be supported by ALSA for others. Thank you for your help, Adam Adam Rosenberg Software Engineer Alcorn McBride Inc. 3300 South Hiawassee Building 105 Orlando, FL 32835 (407) 296 - 5800 ext. 5490

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[alsa-devel] [PATCH] eukrea-tlv320: add support for our i.MX25 board
by Eric Bénard 25 Jun '10

25 Jun '10

* tdm slot has to be configured to get sound working on i.MX25 Signed-off-by: Eric Bénard <eric(a)eukrea.com> --- sound/soc/imx/Kconfig | 2 +- sound/soc/imx/eukrea-tlv320.c | 3 ++- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index fd292f1..fd616ae 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -31,7 +31,7 @@ config SND_SOC_PHYCORE_AC97 config SND_SOC_EUKREA_TLV320 bool "Eukrea TLV320" - depends on MACH_EUKREA_MBIMX27_BASEBOARD + depends on MACH_EUKREA_MBIMX27_BASEBOARD || MACH_EUKREA_MBIMXSD_BASEBOARD select SND_MXC_SOC_SSI select SND_SOC_TLV320AIC23 help diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c index 968380a..45f5e4b 100644 --- a/sound/soc/imx/eukrea-tlv320.c +++ b/sound/soc/imx/eukrea-tlv320.c @@ -60,6 +60,7 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, pr_err("%s: failed setting codec sysclk\n", __func__); return ret; } + snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, SND_SOC_CLOCK_IN); @@ -100,7 +101,7 @@ static int __init eukrea_tlv320_init(void) { int ret; - if (!machine_is_eukrea_cpuimx27()) + if (!machine_is_eukrea_cpuimx27() && !machine_is_eukrea_cpuimx25sd()) /* return happy. We might run on a totally different machine */ return 0; -- 1.6.3.3

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[alsa-devel] [PATCH 1/2] ASoC: uda134x: replace a macro with a value in platform struct.
by Vladimir Zapolskiy 25 Jun '10

25 Jun '10

This change wipes out a hardcoded macro, which enables codec bias level control. Now is_powered_on_standby value shall be used instead. Signed-off-by: Vladimir Zapolskiy <vzapolskiy(a)gmail.com> --- include/sound/uda134x.h | 12 ++++++++++++ sound/soc/codecs/uda134x.c | 21 +++++---------------- 2 files changed, 17 insertions(+), 16 deletions(-) diff --git a/include/sound/uda134x.h b/include/sound/uda134x.h index 509efb0..e475659 100644 --- a/include/sound/uda134x.h +++ b/include/sound/uda134x.h @@ -18,6 +18,18 @@ struct uda134x_platform_data { struct l3_pins l3; void (*power) (int); int model; + /* + ALSA SOC usually puts the device in standby mode when it's not used + for sometime. If you unset is_powered_on_standby the driver will + turn off the ADC/DAC when this callback is invoked and turn it back + on when needed. Unfortunately this will result in a very light bump + (it can be audible only with good earphones). If this bothers you + set is_powered_on_standby, you will have slightly higher power + consumption. Please note that sending the L3 command for ADC is + enough to make the bump, so it doesn't make difference if you + completely take off power from the codec. + */ + int is_powered_on_standby; #define UDA134X_UDA1340 1 #define UDA134X_UDA1341 2 #define UDA134X_UDA1344 3 diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index e8066b6..5c7b054 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -27,19 +27,6 @@ #include "uda134x.h" -#define POWER_OFF_ON_STANDBY 1 -/* - ALSA SOC usually puts the device in standby mode when it's not used - for sometime. If you define POWER_OFF_ON_STANDBY the driver will - turn off the ADC/DAC when this callback is invoked and turn it back - on when needed. Unfortunately this will result in a very light bump - (it can be audible only with good earphones). If this bothers you - just comment this line, you will have slightly higher power - consumption . Please note that sending the L3 command for ADC is - enough to make the bump, so it doesn't make difference if you - completely take off power from the codec. - */ - #define UDA134X_RATES SNDRV_PCM_RATE_8000_48000 #define UDA134X_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE) @@ -559,9 +546,11 @@ static int uda134x_soc_probe(struct platform_device *pdev) codec->num_dai = 1; codec->read = uda134x_read_reg_cache; codec->write = uda134x_write; -#ifdef POWER_OFF_ON_STANDBY - codec->set_bias_level = uda134x_set_bias_level; -#endif + + if (!pd->is_powered_on_standby) { + codec->set_bias_level = uda134x_set_bias_level; + } + INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); -- 1.7.0.4

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Re: [alsa-devel] redirecting audio to another desktop machine
by Hans-Kristian Arntzen 25 Jun '10

25 Jun '10

On 06/21/2010 12:00 PM, alsa-devel-request(a)alsa-project.org wrote: > Date: Mon, 21 Jun 2010 14:57:00 +0530 > From: Viral Mehta<Viral.Mehta(a)lntinfotech.com> > Subject: [alsa-devel] redirecting audio to another desktop machine > To: "alsa-devel(a)alsa-project.org"<alsa-devel(a)alsa-project.org> > Message-ID: > <70376CA23424B34D86F1C7DE6B9973430254343B3E(a)VSHINMSMBX01.vshodc.lntinfotech.com> > > Content-Type: text/plain; charset="us-ascii" > > Hi list, > > I have an embedded device running linux kernel and a minimal filesystem. > > I want to access this device remotely which means I should be able to connect to the device just like we do Remote Desktop over ethernet. > I am able to do this. I am able to send key-event and framebuffer data whenever my desktop machine connects to the device. > > But, I am not able to send audio data on remote side. > I am really not sure which device file (/dev/snd/ ???) I should read and capture the data and should send across.... > Or can I read the playback device ? I think NO. > Do I need to create some kind of virtual sound device ? > > Any pointers/suggestions are appreciated. > > Thanks, > Viral > > > If you want to reroute an ALSA application over the network, you can use output plugins. Afaik, both PulseAudio and RSound provides plugins for this. You generally want to set up a virtual device in .asoundrc or similar. -- Hans-Kristian Arntzen

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[alsa-devel] AndreaElectronics PureAudio USB-SA
by rmfc 25 Jun '10

25 Jun '10

Hello all, I'm trying to use any USB sound card with some DSP processing that allows a microphone array to capture audio and use in speech recognition (mostly applied in noisy environments). The best device that I found so far is AndreaElectronics PureAudio USB-SA but unfortunately it isn't properly supported in Linux/Alsa. The device is recognized as an USB Audio device (snd_usb_audio) and I can record audio in Audacity. The problem is that none of DSP features is explicitly activated in USB configuration so it works almost like any other microphone. I then tried Asus Xonar U1 but the problem is the same, no configuration is done in such devices, so they record audio without enough quality to be used by a speech recognizer (not enough energy, audio DC components when software gain is applied in Sound Preferences...). Is there any way to configure an USB Audio device interfaces? Here's the output of "lsusb" of AndreaElectronics USB-SA sound card. ------------------------------------------------------------ Bus 003 Device 002: ID 08a8:000c Andrea Electronics Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 idVendor 0x08a8 Andrea Electronics idProduct 0x000c bcdDevice 1.00 iManufacturer 3 Andrea Electronics iProduct 1 Andrea PureAudio USB-SA iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 293 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 128 bInCollection 2 baInterfaceNr( 0) 1 baInterfaceNr( 1) 2 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0201 Microphone bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0603 Line Connector bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0605 SPDIF interface bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 6 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 9 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 7 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 8 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 5 (SELECTOR_UNIT) bUnitID 8 bNrInPins 3 baSource( 0) 10 baSource( 1) 11 baSource( 2) 12 iSelector 0 AudioControl Interface Descriptor: bLength 13 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 9 bSourceID 1 bControlSize 2 bmaControls( 0) 0x01 bmaControls( 0) 0x02 Mute Loudness bmaControls( 1) 0x02 bmaControls( 1) 0x00 Volume bmaControls( 2) 0x02 bmaControls( 2) 0x00 Volume iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 10 bSourceID 2 bControlSize 1 bmaControls( 0) 0x01 Mute bmaControls( 1) 0x02 Volume bmaControls( 2) 0x02 Volume iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 11 bSourceID 3 bControlSize 1 bmaControls( 0) 0x01 Mute bmaControls( 1) 0x02 Volume bmaControls( 2) 0x02 Volume iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 12 bSourceID 4 bControlSize 1 bmaControls( 0) 0x01 Mute bmaControls( 1) 0x00 bmaControls( 2) 0x00 iFeature 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 4 Discrete tSamFreq[ 0] 8000 tSamFreq[ 1] 16000 tSamFreq[ 2] 44100 tSamFreq[ 3] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x00c8 1x 200 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 1 Milliseconds wLockDelay 1 Milliseconds Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 4 Discrete tSamFreq[ 0] 8000 tSamFreq[ 1] 16000 tSamFreq[ 2] 44100 tSamFreq[ 3] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x00c8 1x 200 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 No Subclass bInterfaceProtocol 0 None iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.00 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 50 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x87 EP 7 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 bytes bInterval 1 Device Status: 0x0000 (Bus Powered) ------------------------------------------------------------ Renato

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[alsa-devel] [PATCH] ASoC: DaVinci: Added support for stereo I2S
by Raffaele Recalcati 25 Jun '10

25 Jun '10

From: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Added audio playback support with [frame sync master - clock master] mode and with [frame sync master - clock slave]. Clock slave can be important when the external codec need system clock and bit clock synchronized. In the clock master case there is a FIXME message in the source code, because we (Davide and myself) have guessed a frequency of 122000000 that seems the base to be divided. This patch has been developed against the http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git git tree and has been tested on bmx board (similar to dm365 evm, but using uda1345 as external audio codec). Signed-off-by: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Signed-off-by: Davide Bonfanti <davide.bonfanti(a)bticino.it> --- arch/arm/mach-davinci/include/mach/asp.h | 7 ++ sound/soc/davinci/davinci-i2s.c | 141 ++++++++++++++++++++++++++---- 2 files changed, 129 insertions(+), 19 deletions(-) diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 834725f..b1faeb9 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -63,6 +63,13 @@ struct snd_platform_data { unsigned sram_size_playback; unsigned sram_size_capture; + /* + * This define works when both clock and FS are output for the cpu + * and makes clock very fast (FS is not simmetrical, but sampling + * frequency is better approximated + */ + int i2s_fast_clock; + /* McASP specific fields */ int tdm_slots; u8 op_mode; diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index adadcd3..8811d25 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -68,16 +68,23 @@ #define DAVINCI_MCBSP_RCR_RDATDLY(v) ((v) << 16) #define DAVINCI_MCBSP_RCR_RFIG (1 << 18) #define DAVINCI_MCBSP_RCR_RWDLEN2(v) ((v) << 21) +#define DAVINCI_MCBSP_RCR_RFRLEN2(v) ((v) << 24) +#define DAVINCI_MCBSP_RCR_RPHASE (1 << 31) #define DAVINCI_MCBSP_XCR_XWDLEN1(v) ((v) << 5) #define DAVINCI_MCBSP_XCR_XFRLEN1(v) ((v) << 8) #define DAVINCI_MCBSP_XCR_XDATDLY(v) ((v) << 16) #define DAVINCI_MCBSP_XCR_XFIG (1 << 18) #define DAVINCI_MCBSP_XCR_XWDLEN2(v) ((v) << 21) +#define DAVINCI_MCBSP_XCR_XFRLEN2(v) ((v) << 24) +#define DAVINCI_MCBSP_XCR_XPHASE (1 << 31) + +#define CLKGDV(v) (v) /* Bits 0:7 */ #define DAVINCI_MCBSP_SRGR_FWID(v) ((v) << 8) #define DAVINCI_MCBSP_SRGR_FPER(v) ((v) << 16) #define DAVINCI_MCBSP_SRGR_FSGM (1 << 28) +#define DAVINCI_MCBSP_SRGR_CLKSM (1 << 29) #define DAVINCI_MCBSP_PCR_CLKRP (1 << 0) #define DAVINCI_MCBSP_PCR_CLKXP (1 << 1) @@ -144,8 +151,17 @@ struct davinci_mcbsp_dev { * won't end up being swapped because of the underrun. */ unsigned enable_channel_combine:1; + + int i2s_fast_clock; +}; + +struct davinci_mcbsp_data { + unsigned int fmt; + int clk_div; }; +static struct davinci_mcbsp_data mcbsp_data; + static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, int reg, u32 val) { @@ -255,24 +271,27 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int pcr; unsigned int srgr; srgr = DAVINCI_MCBSP_SRGR_FSGM | - DAVINCI_MCBSP_SRGR_FPER(DEFAULT_BITPERSAMPLE * 2 - 1) | - DAVINCI_MCBSP_SRGR_FWID(DEFAULT_BITPERSAMPLE - 1); + DAVINCI_MCBSP_SRGR_FPER(DEFAULT_BITPERSAMPLE * 2 - 1) | + DAVINCI_MCBSP_SRGR_FWID(DEFAULT_BITPERSAMPLE - 1); + /* Attention srgr is updated by hw_params! */ + mcbsp_data.fmt = fmt; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFM: case SND_SOC_DAIFMT_CBS_CFS: /* cpu is master */ pcr = DAVINCI_MCBSP_PCR_FSXM | - DAVINCI_MCBSP_PCR_FSRM | - DAVINCI_MCBSP_PCR_CLKXM | - DAVINCI_MCBSP_PCR_CLKRM; + DAVINCI_MCBSP_PCR_FSRM | + DAVINCI_MCBSP_PCR_CLKXM | + DAVINCI_MCBSP_PCR_CLKRM; break; case SND_SOC_DAIFMT_CBM_CFS: /* McBSP CLKR pin is the input for the Sample Rate Generator. * McBSP FSR and FSX are driven by the Sample Rate Generator. */ pcr = DAVINCI_MCBSP_PCR_SCLKME | - DAVINCI_MCBSP_PCR_FSXM | - DAVINCI_MCBSP_PCR_FSRM; + DAVINCI_MCBSP_PCR_FSXM | + DAVINCI_MCBSP_PCR_FSRM; break; case SND_SOC_DAIFMT_CBM_CFM: /* codec is master */ @@ -372,6 +391,17 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, return 0; } +static int davinci_i2s_dai_set_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct davinci_mcbsp_dev *dev = cpu_dai->private_data; + int srgr; + + mcbsp_data.clk_div = div; + return 0; +} + + static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -380,11 +410,12 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct davinci_pcm_dma_params *dma_params = &dev->dma_params[substream->stream]; struct snd_interval *i = NULL; - int mcbsp_word_length; - unsigned int rcr, xcr, srgr; + int mcbsp_word_length, master; + unsigned int rcr, xcr, srgr, clk_div, freq, framesize; u32 spcr; snd_pcm_format_t fmt; unsigned element_cnt = 1; + struct clk *clk; /* general line settings */ spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); @@ -396,12 +427,60 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); } - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); - srgr = DAVINCI_MCBSP_SRGR_FSGM; - srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1); + master = mcbsp_data.fmt & SND_SOC_DAIFMT_MASTER_MASK; + fmt = params_format(params); + mcbsp_word_length = asp_word_length[fmt]; - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); - srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1); + if (master == SND_SOC_DAIFMT_CBS_CFS) { + clk = clk_get(NULL, "pll1_sysclk6"); + if (clk) + freq = clk_get_rate(clk); + freq = 122000000; /* FIXME ask to Texas */ + if (dev->i2s_fast_clock) { + clk_div = 256; + do { + framesize = (freq / (--clk_div)) / + params->rate_num * + params->rate_den; + } while (((framesize < 33) || (framesize > 4095)) && + (clk_div)); + clk_div--; + srgr = DAVINCI_MCBSP_SRGR_FSGM | + DAVINCI_MCBSP_SRGR_CLKSM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * + 8 - 1); + srgr |= DAVINCI_MCBSP_SRGR_FPER(framesize - 1); + } else { + /* simmetric waveforms */ + srgr = DAVINCI_MCBSP_SRGR_FSGM | + DAVINCI_MCBSP_SRGR_CLKSM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * + 8 - 1); + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * + 16 - 1); + clk_div = freq / (mcbsp_word_length * 16) / + params->rate_num * params->rate_den; + } + clk_div &= 0xFF; + srgr |= clk_div; + } else if (master == SND_SOC_DAIFMT_CBS_CFM) { + /* Clock given on CLKS */ + srgr = DAVINCI_MCBSP_SRGR_FSGM; + clk_div = mcbsp_data.clk_div - 1; + srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * 8 - 1); + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * 16 - 1); + clk_div &= 0xFF; + srgr |= clk_div; + } else { + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); + srgr = DAVINCI_MCBSP_SRGR_FSGM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1); + pr_debug("%s - %d FWID set: re-read srgr = %X\n", + __func__, __LINE__, snd_interval_value(i) - 1); + + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); + srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1); + } davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, srgr); rcr = DAVINCI_MCBSP_RCR_RFIG; @@ -426,22 +505,45 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, element_cnt = 1; fmt = double_fmt[fmt]; } + if (master == SND_SOC_DAIFMT_CBS_CFS || + master == SND_SOC_DAIFMT_CBS_CFM) { + rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(0); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(0); + rcr |= DAVINCI_MCBSP_RCR_RPHASE; + xcr |= DAVINCI_MCBSP_XCR_XPHASE; + } else { + /* FIXME ask to Texas */ + rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(element_cnt - 1); + } } dma_params->acnt = dma_params->data_type = data_type[fmt]; dma_params->fifo_level = 0; mcbsp_word_length = asp_word_length[fmt]; - rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); - xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); + + if (master == SND_SOC_DAIFMT_CBS_CFS || + master == SND_SOC_DAIFMT_CBS_CFM) { + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(0); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(0); + } else { + /* FIXME ask to Texas */ + 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); + DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length); xcr |= DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) | - DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length); + DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, xcr); else davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, rcr); + + pr_debug("%s - %d srgr=%X\n", __func__, __LINE__, srgr); + pr_debug("%s - %d xcr=%X\n", __func__, __LINE__, xcr); + pr_debug("%s - %d rcr=%X\n", __func__, __LINE__, rcr); return 0; } @@ -500,7 +602,7 @@ static struct snd_soc_dai_ops davinci_i2s_dai_ops = { .trigger = davinci_i2s_trigger, .hw_params = davinci_i2s_hw_params, .set_fmt = davinci_i2s_set_dai_fmt, - + .set_clkdiv = davinci_i2s_dai_set_clkdiv, }; struct snd_soc_dai davinci_i2s_dai = { @@ -548,6 +650,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) } if (pdata) { dev->enable_channel_combine = pdata->enable_channel_combine; + dev->i2s_fast_clock = pdata->i2s_fast_clock; dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].sram_size = pdata->sram_size_playback; dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = -- 1.7.0.4

3 3

[alsa-devel] Compilation error
by lars.bjorndal＠broadpark.no 25 Jun '10

25 Jun '10

I get the following output while trying to compile the alsa driver on my Fedora 12 system. It worked well with the previous kernel version. # ./alsa-compile.sh --install File environment.base is cached. > cd /tmp/alsa-compile-script File environment.compile is cached. alsa-driver tree alsa-driver-1.0.23.31.g2379a.273.g5b97d is present. Reusing it. Use './alsa-compile.sh --clean=alsa-driver' command to refetch and rebuild. Sources unpacked to alsa-driver-1.0.23.31.g2379a.273.g5b97d > cd alsa-driver-1.0.23.31.g2379a.273.g5b97d Running make: make dep make[1]: Entering directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d' make[2]: Entering directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/include' make -C sound prepare make[3]: Entering directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/include/sound' make prepare2 make[4]: Entering directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/include/sound' make[4]: Nothing to be done for `prepare2'. make[4]: Leaving directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/include/sound' make[3]: Leaving directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/include/sound' make[2]: Leaving directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/include' ... make[2]: Entering directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/misc' make[2]: Leaving directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/misc' make[1]: Leaving directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d' make -C /lib/modules/2.6.32.14-127.fc12.i686.PAE/build SUBDIRS=/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d CPP="gcc -E" CC="gcc" modules make[1]: Entering directory `/usr/src/kernels/2.6.32.14-127.fc12.i686.PAE' CC [M] /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/ua101.o In file included from /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/ua101.c:2: /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/../../alsa-kernel/usb/misc/ua101.c: In function 'alloc_stream_buffers': /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/../../alsa-kernel/usb/misc/ua101.c:1071: error: implicit declaration of function 'usb_alloc_coherent' /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/../../alsa-kernel/usb/misc/ua101.c:1072: warning: assignment makes pointer from integer without a cast /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/../../alsa-kernel/usb/misc/ua101.c: In function 'free_stream_buffers': /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/../../alsa-kernel/usb/misc/ua101.c:1092: error: implicit declaration of function 'usb_free_coherent' /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/../../alsa-kernel/usb/misc/ua101.c: In function 'ua101_card_free': /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/../../alsa-kernel/usb/misc/ua101.c:1172: warning: unused variable 'ua' make[4]: *** [/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/ua101.o] Error 1 make[3]: *** [/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc] Error 2 make[2]: *** [/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb] Error 2 make[1]: *** [_module_/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.32.14-127.fc12.i686.PAE' make: *** [compile] Error 2 Working directory /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d. Compilation of alsa-driver failed. Report this problem to the <alsa-devel(a)alsa-project.org> mailing list. Lars

1 0

[alsa-devel] Verifying mixer dB data/Invalid dB data from USB cards, especially Aureon 5.1 MkII
by Lennart Poettering 25 Jun '10

25 Jun '10

Heya, a while back I wrote a little tool "dbmeasure" which can be used to determine dB attenuation data for alsa mixer controls. I have now extended this little dB tools suite to include a tiny tool "dbverify" that can be used to verify the correctness of existing dB data exposed by the drivers. It should be relatively easy to use that tool, suitable even for non-guru folks who want to debug the dB information from alsa. http://git.0pointer.de/?p=dbmeasure.git git://git.0pointer.de/dbmeasure.git Just run "make" in a git checkout and then try: $ ./dbverify Aureon51MkII Master 30 200 That will verify the dB data of the "Master" control of the sound card that goes by the id "Aureon51MkII". It will compare the discrete volume steps 30 and 200: first it will play a full amplitude sine wave at mixer volume step 30, and then will change to mixer volume step 200, but attenuate the sine wave in software compensating for the volume change. So once you hear the sine wave attenuated by the hw, and once by the software. If the card's dB data is correct both sines should have the same volume. Now, playing around with this, I could verify what I already mentioned to Jarsolav elsewhere earlier: it seems that particularly USB cards seem to expose invalid dB data (at least all mine do). For example, for the above mentioned Aureon 5.1 it's really *way* off -- if you posess that card just run the line above, and listen to the difference. It's really bad. OTOH the dB data of the HDA cards I own is pretty accurate as it seems, at least I was unable to decide just by listening which sine wave was attenuated by sw, and which by hw. Getting back to the invalid dB data from the USB cards: the question is whether the USB descriptor data is badly parsed, and the dB mismatch hence systematic for USB cards, or if these cards are just crappy and include invalid data? In which case I wonder what we could do about that? The Aureon's dB range already looks suspicous, since both the max and the min dB value are < 0 (-47.69 to -1.97), so maybe we could add some heuristics to filter out data that already looks suspicous? Invalid dB data from the driver is a real problem for the "flat volume" logic in PA. We basically allow each app to control the full hw volume range individually, and then set the hw volume to the max of what all apps wanted and attenuate the other streams accordingly. On the Aureon this doesn't work at all, since the attenuation of the streams is miscalculated due to the invalid dB data. Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net http://0pointer.net/lennart/ GnuPG 0x1A015CC4

8 93

[alsa-devel] [RFC 5/5] ASoC multi-component support : jack
by Liam Girdwood 24 Jun '10

24 Jun '10

Cache configuration is now stored as part of the codec driver structure. Signed-off-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- sound/soc/soc-cache.c | 34 +++++++++++++++++----------------- 1 files changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 472af38..83cd8ed 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -19,7 +19,7 @@ static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, unsigned int reg) { u16 *cache = codec->reg_cache; - if (reg >= codec->reg_cache_size) + if (reg >= codec->driver->reg_cache_size) return -1; return cache[reg]; } @@ -31,12 +31,12 @@ static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, u8 data[2]; int ret; - BUG_ON(codec->volatile_register); + BUG_ON(codec->driver->volatile_register); data[0] = (reg << 4) | ((value >> 8) & 0x000f); data[1] = value & 0x00ff; - if (reg < codec->reg_cache_size) + if (reg < codec->driver->reg_cache_size) cache[reg] = value; if (codec->cache_only) { @@ -89,7 +89,7 @@ static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec, unsigned int reg) { u16 *cache = codec->reg_cache; - if (reg >= codec->reg_cache_size) + if (reg >= codec->driver->reg_cache_size) return -1; return cache[reg]; } @@ -101,12 +101,12 @@ static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, u8 data[2]; int ret; - BUG_ON(codec->volatile_register); + BUG_ON(codec->driver->volatile_register); data[0] = (reg << 1) | ((value >> 8) & 0x0001); data[1] = value & 0x00ff; - if (reg < codec->reg_cache_size) + if (reg < codec->driver->reg_cache_size) cache[reg] = value; if (codec->cache_only) { @@ -161,13 +161,13 @@ static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, u8 *cache = codec->reg_cache; u8 data[2]; - BUG_ON(codec->volatile_register); + BUG_ON(codec->driver->volatile_register); reg &= 0xff; data[0] = reg; data[1] = value & 0xff; - if (reg < codec->reg_cache_size) + if (reg < codec->driver->reg_cache_size) cache[reg] = value; if (codec->cache_only) { @@ -188,7 +188,7 @@ static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec, { u8 *cache = codec->reg_cache; reg &= 0xff; - if (reg >= codec->reg_cache_size) + if (reg >= codec->driver->reg_cache_size) return -1; return cache[reg]; } @@ -224,7 +224,7 @@ static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec, { u16 *cache = codec->reg_cache; - if (reg >= codec->reg_cache_size || + if (reg >= codec->driver->reg_cache_size || snd_soc_codec_volatile_register(codec, reg)) { if (codec->cache_only) return -EINVAL; @@ -343,7 +343,7 @@ static unsigned int snd_soc_16_8_read(struct snd_soc_codec *codec, u16 *cache = codec->reg_cache; reg &= 0xff; - if (reg >= codec->reg_cache_size) + if (reg >= codec->driver->reg_cache_size) return -1; return cache[reg]; } @@ -355,14 +355,14 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, u8 data[3]; int ret; - BUG_ON(codec->volatile_register); + BUG_ON(codec->driver->volatile_register); data[0] = (reg >> 8) & 0xff; data[1] = reg & 0xff; data[2] = value; reg &= 0xff; - if (reg < codec->reg_cache_size) + if (reg < codec->driver->reg_cache_size) cache[reg] = value; if (codec->cache_only) { @@ -451,7 +451,7 @@ static unsigned int snd_soc_16_16_read(struct snd_soc_codec *codec, { u16 *cache = codec->reg_cache; - if (reg >= codec->reg_cache_size || + if (reg >= codec->driver->reg_cache_size || snd_soc_codec_volatile_register(codec, reg)) { if (codec->cache_only) return -EINVAL; @@ -474,7 +474,7 @@ static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, data[2] = (value >> 8) & 0xff; data[3] = value & 0xff; - if (reg < codec->reg_cache_size) + if (reg < codec->driver->reg_cache_size) cache[reg] = value; if (codec->cache_only) { @@ -571,8 +571,8 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, return -EINVAL; } - codec->write = io_types[i].write; - codec->read = io_types[i].read; + codec->driver->write = io_types[i].write; + codec->driver->read = io_types[i].read; switch (control) { case SND_SOC_CUSTOM: -- 1.7.0.4

1 0

[alsa-devel] [RFC 4/5] ASoC multi-component support : jack
by Liam Girdwood 24 Jun '10

24 Jun '10

Jack interface is per codec rather than per card in order to support sound cards with more than one codec. Signed-off-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- sound/soc/soc-jack.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 29159e1..1dfc4fd 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -32,14 +32,14 @@ * Returns zero if successful, or a negative error code on failure. * On success jack will be initialised. */ -int snd_soc_jack_new(struct snd_soc_card *card, const char *id, int type, +int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type, struct snd_soc_jack *jack) { - jack->card = card; + jack->codec = codec; INIT_LIST_HEAD(&jack->pins); BLOCKING_INIT_NOTIFIER_HEAD(&jack->notifier); - return snd_jack_new(card->codec->card, id, type, &jack->jack); + return snd_jack_new(codec->snd_card, id, type, &jack->jack); } EXPORT_SYMBOL_GPL(snd_soc_jack_new); @@ -67,7 +67,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) if (!jack) return; - codec = jack->card->codec; + codec = jack->codec; mutex_lock(&codec->mutex); @@ -268,7 +268,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, ret = request_irq(gpio_to_irq(gpios[i].gpio), gpio_handler, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING, - jack->card->dev->driver->name, + jack->codec->dev->driver->name, &gpios[i]); if (ret) goto err; -- 1.7.0.4

1 0