[alsa-devel] SoC on PXA27x with two codecs
Liam Girdwood
lrg at slimlogic.co.uk
Tue Feb 2 11:37:01 CET 2010
On Tue, 2010-02-02 at 10:58 +0100, Rodolfo Giometti wrote:
> Hello,
>
> on my custom board I have a PXA27x CPU and 2 codecs: one connected as
> primary codec and one connected as secondary codec.
>
> I modify the SoC support in order to allow defining two audio cards,
> one to manage the first codec and one to manage the second one.
>
> Here the SoC schema:
>
> static struct snd_soc_dai_link bt270p_dai[] = {
> {
> .name = "AC97 HiFi",
> .stream_name = "audio out",
> .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
> .codec_dai = &cs4299_dai[CS4299_DAI_AC97_HIFI],
> .init = bt270_ac97_init,
> .ops = &bt270_ops,
> },
> {
> .name = "AC97 Aux",
> .stream_name = "AC97 Aux",
> .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
> .codec_dai = &cs4299_dai[CS4299_DAI_AC97_AUX],
> .init = bt270_ac97_init,
> .ops = &bt270_ops,
> },
> };
>
> static struct snd_soc_dai_link bt270s_dai[] = {
> {
> .name = "AC97 HiFi",
> .stream_name = "speakers",
> .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
> .codec_dai = &cs4299_dai[CS4299_DAI_AC97_HIFI],
> .init = bt270_ac97_init,
> .ops = &bt270_ops,
> },
> };
>
> static struct snd_soc_card bt270_asoc[] = {
> {
> .name = "BT270 primary",
> .platform = &pxa2xx_soc_platform,
> .dai_link = bt270p_dai,
> .num_links = ARRAY_SIZE(bt270p_dai),
> },
> {
> .name = "BT270 secondary",
> .platform = &pxa2xx_soc_platform,
> .dai_link = bt270s_dai,
> .num_links = ARRAY_SIZE(bt270s_dai),
> },
> };
>
> static struct cs4299_data cs4299_data[] = {
> {
> .ac97_line = 0,
> },
> {
> .ac97_line = 1,
> },
> };
>
> static struct snd_soc_device bt270_snd_devdata[2] = {
> {
> .card = &bt270_asoc[0],
> .codec_dev = &soc_codec_dev_cs4299,
> .codec_data = &cs4299_data[0],
> },
> {
> .card = &bt270_asoc[1],
> .codec_dev = &soc_codec_dev_cs4299,
> .codec_data = &cs4299_data[1],
> },
> };
>
> static struct platform_device *bt270_snd_device[2];
>
> I'm using the ac97_line parameters into codec's code in order to
> select the primary and the second codecs as follow:
>
> static int cs4299_soc_probe(struct platform_device *pdev)
> {
> struct snd_soc_device *socdev = platform_get_drvdata(pdev);
> struct cs4299_data *data = (struct cs4299_data *) socdev->codec_data;
> struct snd_soc_codec *codec;
> int ac97_line;
> int ret = 0;
>
> printk(KERN_INFO "CS4299 SoC Audio Codec %s\n", CS4299_VERSION);
>
> /* Check specific codec data or set default if not defined */
> ac97_line = 0;
> if (data)
> ac97_line = data->ac97_line;
>
> socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec),
> GFP_KERNEL);
> if (socdev->card->codec == NULL)
> return -ENOMEM;
> codec = socdev->card->codec;
> mutex_init(&codec->mutex);
>
> codec->reg_cache = kmemdup(cs4299_reg, sizeof(cs4299_reg), GFP_KERNEL);
>
> if (codec->reg_cache == NULL) {
> ret = -ENOMEM;
> goto cache_err;
> }
> codec->reg_cache_size = sizeof(cs4299_reg);
> codec->reg_cache_step = 2;
>
> codec->name = "CS4299";
> codec->id = ac97_line;
> codec->owner = THIS_MODULE;
> codec->dai = cs4299_dai;
> codec->num_dai = ARRAY_SIZE(cs4299_dai);
> codec->write = ac97_write;
> codec->read = ac97_read;
> INIT_LIST_HEAD(&codec->dapm_widgets);
> INIT_LIST_HEAD(&codec->dapm_paths);
>
> ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, ac97_line);
> if (ret < 0) {
> printk(KERN_ERR "cs4299: failed to register AC97
> codec\n");
> goto codec_err;
> }
>
> /* register pcms */
> ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
> if (ret < 0)
> goto pcm_err;
>
> ret = cs4299_reset(codec, 0);
> if (ret < 0) {
> printk(KERN_ERR "Failed to reset CS4299: AC97 link
> error\n");
> goto reset_err;
> }
>
> snd_soc_add_controls(codec, cs4299_snd_ac97_controls,
> ARRAY_SIZE(cs4299_snd_ac97_controls));
> cs4299_add_widgets(codec);
> ret = snd_soc_init_card(socdev);
> if (ret < 0) {
> printk(KERN_ERR "cs4299: failed to register card\n");
> goto reset_err;
> }
>
> return 0;
>
> Everything is working ok but the PCM play/record. In fact I can manage
> both codecs' registers by using:
>
> alsamixer -c 0 and
> alsamixer -c 1
>
> but when I try to play an MP3 file I can hear the music only on the
> second codec no matter which audio card I select for playing.
>
> I tested the hardware by defining just one codec at time, and doing in
> this manner the music goes to the right codec.
>
> Any suggeston? :)
What about the debug output from the DMA configuration (wrt AC97 slot
FIFOs) I suggested yesterday ?
We really need to confirm the AC97 slots being used and whether each
codec is listening on these slots.
Liam
--
Freelance Developer, SlimLogic Ltd
ASoC and Voltage Regulator Maintainer.
http://www.slimlogic.co.uk
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