[alsa-devel] [PATCH] New ASoC Drivers for ADI AD1938 codec
宋宝华
21cnbao at gmail.com
Mon Jun 22 05:08:27 CEST 2009
Hi Mark,
***For the new DAI format
According to I2S spec, it doesn't definite a I2S with TDM as a standard I2S.
http://www.nxp.com/acrobat_download/various/I2SBUS.pdf
It looks like you are admitting this kind of timing into I2S DAI too:
http://i3.6.cn/cvbnm/8f/3d/08/268a4560e0daa1b41d69b82419da06e1.jpg
I think I can follow it too.
Due to my test boards, at present, the AD1938 is working in and supporting
TDM timing like the diagram:
http://i3.6.cn/cvbnm/2f/e2/f2/03ae2b51c4e90749972e70bf887f926f.jpg
It looks like DSP mode with TDM, so can I path related codes into
SND_SOC_DAIFMT_DSP switch?
***For volume controls based on stereo pairs
Even though DAC1-DAC8 are named as DACL1,DACR1, DACL2,DACR2..., but the
DACLx and DACRx are not always in a pair, in fact, they are independent. As
a codec supporting 8 channels, it can be configed into 2, 2.1, 4.1, 5.1,
6.1, 7.1, how to handle the pairs?
Thanks
Barry
2009/6/19 Mark Brown <broonie at opensource.wolfsonmicro.com>
> On Fri, Jun 19, 2009 at 05:28:15PM +0800, Barry Song wrote:
> > 1. add AD1938 codec driver (codec)
> > 2. add blackfin SPORT-TDM DAI and PCM driver (platform)
> > 3. add bf5xx board with AD1938 driver (machine)
>
> As Liam said you really need to submit this as a patch series rather
> than as a single big patch - as your commit log here indicates you've
> got several different things going on here.
>
> > +++ b/include/sound/soc-dai.h
> > @@ -30,6 +30,7 @@ struct snd_pcm_substream;
> > #define SND_SOC_DAIFMT_DSP_A 3 /* L data msb after FRM LRC */
> > #define SND_SOC_DAIFMT_DSP_B 4 /* L data msb during FRM LRC */
> > #define SND_SOC_DAIFMT_AC97 5 /* AC97 */
> > +#define SND_SOC_DAIFMT_SPORT_TDM 6 /* SPORT TDM for ADI parts */
>
> If you're going to add a new DAI format that really needs more
> explanation than this explaining what the DAI format is. It'd be very
> surprising to see hardware needing a new format.
>
> Looking at the datasheet for the ad1938 it appears that the actual
> format here is just normal I2S with TDM. This does not need a new DAI
> format or new CPU DAI, you just need to add suport for TDM to the
> Blackfin I2S driver. The format is fairly standard and implemented by a
> number of other devices.
>
> See set_tdm_slot() for setting up the higher channel counts - there's
> some ongoing revisions to that API so you'll want to also ensure that
> the code is set up so that it can cope with specification of the sample
> width for each slot in set_tdm_slot().
>
> Given this I've only looked at the CODEC driver below.
>
> > diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c
> > new file mode 100644
> > index 0000000..9aa78e1
> > --- /dev/null
> > +++ b/sound/soc/codecs/ad1938.c
>
> > + *
> > + * Revision history
> > + * 4 June 2009 Initial version.
>
> Don't include this, git provides code history for us.
>
> > +struct snd_soc_device *ad1938_socdev;
> > +
> > +/* dac de-emphasis enum control */
> > +static const char *ad1938_deemp[] = {"flat", "48kHz", "44.1kHz",
> "32kHz"};
>
> For consistency with other drivers "flat" should be "None".
>
> > +/* AD1938 volume/mute/de-emphasis etc. controls */
> > +static const struct snd_kcontrol_new ad1938_snd_controls[] = {
> > + /* DAC volume control */
> > + SOC_SINGLE("DAC L1 Volume", AD1938_DAC_L1_VOL, 0, 0xFF, 1),
> > + SOC_SINGLE("DAC R1 Volume", AD1938_DAC_R1_VOL, 0, 0xFF, 1),
>
> These (and the other stereo pairs below) should be SOC_DOUBLE_R(). This
> allows ALSA to represent them as stereo controls to applications rather
> than as two separate controls. You should also provide TLV information
> so actually SOC_DOUBLE_R_TLV() if possible.
>
> > + /* DAC mute control */
> > + SOC_SINGLE("DAC L1 Switch", AD1938_DAC_CHNL_MUTE, 0, 1, 1),
> > + SOC_SINGLE("DAC R1 Switch", AD1938_DAC_CHNL_MUTE, 1, 1, 1),
>
> These should be stereo controls too - SOC_DOUBLE() since they're in the
> same register.
>
> > + /* ADC mute control */
> > + SOC_SINGLE("ADC L1 Switch", AD1938_ADC_CTRL0, ADC0_MUTE, 1, 1),
> > + SOC_SINGLE("ADC R1 Switch", AD1938_ADC_CTRL0, ADC1_MUTE, 1, 1),
>
> These too.
>
> > + /* DAC de-emphasis */
> > + SOC_ENUM("Playback Deemphasis", ad1938_enum[0]),
>
> Don't put your enums in an array, use named variables for them. This
> makes drivers easier to maintian when you get a lot of enums.
>
> > +static int ad1938_add_controls(struct snd_soc_codec *codec)
> > +{
> > + int err, i;
> > +
> > + for (i = 0; i < ARRAY_SIZE(ad1938_snd_controls); i++) {
> > + err = snd_ctl_add(codec->card,
> > + snd_soc_cnew(&ad1938_snd_controls[i],
> codec, NULL));
>
> Use snd_soc_add_controls() here - you can replace the entire function
> with a call to that.
>
> > +/* dac/adc/pll poweron/off functions */
> > +static int ad1938_dac_powerctrl(struct snd_soc_codec *codec, int cmd)
> > +{
> > + int reg;
> > +
> > + reg = codec->read(codec, AD1938_DAC_CTRL0);
> > + if (cmd)
> > + reg &= ~DAC_POWERDOWN;
> > + else
> > + reg |= DAC_POWERDOWN;
> > + codec->write(codec, AD1938_DAC_CTRL0, reg);
>
> This should be handled by DAPM - either have a single DAC widget
> representing all the channels (since you don't appear to have
> independant control anyway) or have a bunch of dummy DAC widgets and a
> supply widget representing the actual power control. The same thing
> applies to the ADCs.
>
> > +static int ad1938_set_pll(struct snd_soc_dai *codec_dai,
> > + int pll_id, unsigned int freq_in, unsigned int freq_out)
> > +{
> > + struct snd_soc_codec *codec = codec_dai->codec;
> > +
> > + if (freq_out)
> > + ad1938_pll_powerctrl(codec, 1);
> > + else {
> > + /* playing while recording, framework will poweroff-poweron
> pll redundantly */
> > + if ((!codec_dai->capture.active) &&
> (!codec_dai->playback.active))
> > + ad1938_pll_powerctrl(codec, 0);
> > + }
>
> Hrm. This appears to completely ignore the frequencies supplied for the
> PLL and just provide power control. I suspect that you can just handle
> the PLL as a SND_SOC_DAPM_SUPPLY(), there seems to be no need to expose
> the set_pll() operation and make machine drivers call it given that
> there isn't any frequency configuration going on.
>
> > +static int ad1938_mute(struct snd_soc_dai *dai, int mute)
> > +{
> > + struct snd_soc_codec *codec = dai->codec;
> > +
> > + if (!mute)
> > + codec->write(codec, AD1938_DAC_CHNL_MUTE, 0);
> > + else
> > + codec->write(codec, AD1938_DAC_CHNL_MUTE, 0xff);
> > +
> > + return 0;
> > +}
>
> This isn't going to play well with the explicit mute controls you've got
> above - it's writing to the same register bits without any coordination.
> One or the other set of controls ought to be removed.
>
> > +static int ad1938_tdm_set(struct snd_soc_codec *codec)
> > +{
> > + codec->write(codec, AD1938_DAC_CTRL0, (codec->read(codec,
> AD1938_DAC_CTRL0) &
> > + (~DAC_SERFMT_MASK)) | DAC_SERFMT_TDM);
> > + codec->write(codec, AD1938_DAC_CTRL1, 0x84); /* invert bclk,
> 256bclk/frame, latch in mid */
> > + codec->write(codec, AD1938_ADC_CTRL1, 0x43); /* sata delay=1, adc
> aux mode */
> > + codec->write(codec, AD1938_ADC_CTRL2, 0x6F); /* left high, driver
> on rising edge */
> > +
> > + return 0;
> > +}
>
> If you use set_tdm_slot() then the BCLK/frame ratio will be set by that.
>
> Inversion of BCLK (and any other clocks) should be handled by the
> set_dai_fmt() operation based on the machine driver request rather than
> done unconditionally.
>
> > + /* bit size */
> > + switch (params_format(params)) {
> > + case SNDRV_PCM_FORMAT_S16_LE:
> > + word_len = 3;
> > + break;
>
> Once you implement set_tdm_slot() you should allow the word length to be
> configured there if it's called or otherwise keep this code here - see
> Daniel Ribeiro's patche "change set_tdm_slot api to allow slot_width
> override" posted to the ALSA list this week.
>
> > +static int __devinit ad1938_spi_probe(struct spi_device *spi)
> > +{
> > + spi->dev.power.power_state = PMSG_ON;
> > + ad1938_socdev->card->codec->control_data = spi;
> > +
> > + return 0;
> > +}
> > +
> > +static int __devexit ad1938_spi_remove(struct spi_device *spi)
> > +{
> > + return 0;
> > +}
>
> Your device probing should all be restructured so that the SPI device
> for the CODEC is registered as any other SPI device rather than being
> set up as part of probing the ASoC device. See the wm8731 driver for
> an example of doing this for a SPI device.
>
> This will require that the arch code for any systems with the ad1938
> do the setup of the device.
>
> > + .name = "AD1938",
> > + .playback = {
> > + .stream_name = "Playback",
> > + .channels_min = 2,
> > + .channels_max = 8,
> > + .rates = SNDRV_PCM_RATE_48000,
> > + .formats = SNDRV_PCM_FMTBIT_S32_LE |
> SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |
> SNDRV_PCM_FMTBIT_S24_LE, },
>
> Please keep your lines to under 80 columns.
>
> > +#define AD1938_PLL_CLK_CTRL0 0
> > +#define PLL_POWERDOWN 0x01
> > +#define AD1938_PLL_CLK_CTRL1 1
> > +#define AD1938_DAC_CTRL0 2
> > +#define DAC_POWERDOWN 0x01
> > +#define DAC_SERFMT_MASK 0xC0
> > +#define DAC_SERFMT_STEREO (0 << 6)
> > +#define DAC_SERFMT_TDM (1 << 6)
>
> These defines need namespacing if they're going to appear in the headers
> - everything should have the AD1938_ prefix.
>
--
宋宝华 21cnbao at 21cn.com
http://21cnbao.blog.51cto.com
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