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authorLinus Torvalds <torvalds@linux-foundation.org>2012-12-13 11:51:23 -0800
committerLinus Torvalds <torvalds@linux-foundation.org>2012-12-13 11:51:23 -0800
commit046e7d685bc370fd4c879ab6635ad3f69e6673d1 (patch)
tree36b981f8d1f2bfd348c1479acbe3a9426d35c377 /sound/soc/codecs/si476x.c
parentfe504c5c745aeb767d978fbedeb94775fd4cb69c (diff)
parent6eb827d23577a4efec2b10a9c4cc9ded268a1d1c (diff)
downloadblackbird-op-linux-046e7d685bc370fd4c879ab6635ad3f69e6673d1.tar.gz
blackbird-op-linux-046e7d685bc370fd4c879ab6635ad3f69e6673d1.zip
Merge tag 'sound-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai: "This update contains a fairly wide range of changes all over in sound subdirectory, mainly because of UAPI header moves by David and __dev* annotation removals by Bill. Other highlights are: - Introduced the support for wallclock timestamps in ALSA PCM core - Add the poll loop implementation for HD-audio jack detection - Yet more VGA-switcheroo fixes for HD-audio - New VIA HD-audio codec support - More fixes on resource management in USB audio and MIDI drivers - More quirks for USB-audio ASUS Xonar U3, Reloop Play, Focusrite, Roland VG-99, etc - Add support for FastTrack C400 usb-audio - Clean ups in many drivers regarding firmware loading - Add PSC724 Ultiimate Edge support to ice1712 - A few hdspm driver updates - New Stanton SCS.1d/1m FireWire driver - Standardisation of the logging in ASoC codes - DT and dmaengine support for ASoC Atmel - Support for Wolfson ADSP cores - New drivers for Freescale/iVeia P1022 and Maxim MAX98090 - Lots of other ASoC driver fixes and developments" Fix up trivial conflicts. And go out on a limb and assume the dts file 'status' field of one of the conflicting things was supposed to be "disabled", not "disable" like in pretty much all other cases. * tag 'sound-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (341 commits) ALSA: hda - Move runtime PM check to runtime_idle callback ALSA: hda - Add stereo-dmic fixup for Acer Aspire One 522 ALSA: hda - Avoid doubly suspend after vga switcheroo ALSA: usb-audio: Enable S/PDIF on the ASUS Xonar U3 ALSA: hda - Check validity of CORB/RIRB WP reads ALSA: hda - use usleep_range in link reset and change timeout check ALSA: HDA: VIA: Add support for codec VT1808. ALSA: HDA: VIA Add support for codec VT1705CF. ASoC: codecs: remove __dev* attributes ASoC: utils: remove __dev* attributes ASoC: ux500: remove __dev* attributes ASoC: txx9: remove __dev* attributes ASoC: tegra: remove __dev* attributes ASoC: spear: remove __dev* attributes ASoC: sh: remove __dev* attributes ASoC: s6000: remove __dev* attributes ASoC: OMAP: remove __dev* attributes ASoC: nuc900: remove __dev* attributes ASoC: mxs: remove __dev* attributes ASoC: kirkwood: remove __dev* attributes ...
Diffstat (limited to 'sound/soc/codecs/si476x.c')
-rw-r--r--sound/soc/codecs/si476x.c255
1 files changed, 255 insertions, 0 deletions
diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c
new file mode 100644
index 000000000000..f2d61a187830
--- /dev/null
+++ b/sound/soc/codecs/si476x.c
@@ -0,0 +1,255 @@
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+
+#include <linux/i2c.h>
+
+#include <linux/mfd/si476x-core.h>
+
+enum si476x_audio_registers {
+ SI476X_DIGITAL_IO_OUTPUT_FORMAT = 0x0203,
+ SI476X_DIGITAL_IO_OUTPUT_SAMPLE_RATE = 0x0202,
+};
+
+enum si476x_digital_io_output_format {
+ SI476X_DIGITAL_IO_SLOT_SIZE_SHIFT = 11,
+ SI476X_DIGITAL_IO_SAMPLE_SIZE_SHIFT = 8,
+};
+
+#define SI476X_DIGITAL_IO_OUTPUT_WIDTH_MASK ((0b111 << SI476X_DIGITAL_IO_SLOT_SIZE_SHIFT) | \
+ (0b111 << SI476X_DIGITAL_IO_SAMPLE_SIZE_SHIFT))
+#define SI476X_DIGITAL_IO_OUTPUT_FORMAT_MASK (0b1111110)
+
+enum si476x_daudio_formats {
+ SI476X_DAUDIO_MODE_I2S = (0x0 << 1),
+ SI476X_DAUDIO_MODE_DSP_A = (0x6 << 1),
+ SI476X_DAUDIO_MODE_DSP_B = (0x7 << 1),
+ SI476X_DAUDIO_MODE_LEFT_J = (0x8 << 1),
+ SI476X_DAUDIO_MODE_RIGHT_J = (0x9 << 1),
+
+ SI476X_DAUDIO_MODE_IB = (1 << 5),
+ SI476X_DAUDIO_MODE_IF = (1 << 6),
+};
+
+enum si476x_pcm_format {
+ SI476X_PCM_FORMAT_S8 = 2,
+ SI476X_PCM_FORMAT_S16_LE = 4,
+ SI476X_PCM_FORMAT_S20_3LE = 5,
+ SI476X_PCM_FORMAT_S24_LE = 6,
+};
+
+static unsigned int si476x_codec_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ int err;
+ struct si476x_core *core = codec->control_data;
+
+ si476x_core_lock(core);
+ err = si476x_core_cmd_get_property(core, reg);
+ si476x_core_unlock(core);
+
+ return err;
+}
+
+static int si476x_codec_write(struct snd_soc_codec *codec,
+ unsigned int reg, unsigned int val)
+{
+ int err;
+ struct si476x_core *core = codec->control_data;
+
+ si476x_core_lock(core);
+ err = si476x_core_cmd_set_property(core, reg, val);
+ si476x_core_unlock(core);
+
+ return err;
+}
+
+static int si476x_codec_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ int err;
+ u16 format = 0;
+
+ if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS)
+ return -EINVAL;
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_DSP_A:
+ format |= SI476X_DAUDIO_MODE_DSP_A;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ format |= SI476X_DAUDIO_MODE_DSP_B;
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ format |= SI476X_DAUDIO_MODE_I2S;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ format |= SI476X_DAUDIO_MODE_RIGHT_J;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ format |= SI476X_DAUDIO_MODE_LEFT_J;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_DSP_A:
+ case SND_SOC_DAIFMT_DSP_B:
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ format |= SI476X_DAUDIO_MODE_IB;
+ break;
+ default:
+ return -EINVAL;
+ }
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ case SND_SOC_DAIFMT_RIGHT_J:
+ case SND_SOC_DAIFMT_LEFT_J:
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ format |= SI476X_DAUDIO_MODE_IB |
+ SI476X_DAUDIO_MODE_IF;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ format |= SI476X_DAUDIO_MODE_IB;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ format |= SI476X_DAUDIO_MODE_IF;
+ break;
+ default:
+ return -EINVAL;
+ }
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ err = snd_soc_update_bits(codec_dai->codec, SI476X_DIGITAL_IO_OUTPUT_FORMAT,
+ SI476X_DIGITAL_IO_OUTPUT_FORMAT_MASK,
+ format);
+ if (err < 0) {
+ dev_err(codec_dai->codec->dev, "Failed to set output format\n");
+ return err;
+ }
+
+ return 0;
+}
+
+static int si476x_codec_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ int rate, width, err;
+
+ rate = params_rate(params);
+ if (rate < 32000 || rate > 48000) {
+ dev_err(dai->codec->dev, "Rate: %d is not supported\n", rate);
+ return -EINVAL;
+ }
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S8:
+ width = SI476X_PCM_FORMAT_S8;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ width = SI476X_PCM_FORMAT_S16_LE;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ width = SI476X_PCM_FORMAT_S20_3LE;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ width = SI476X_PCM_FORMAT_S24_LE;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ err = snd_soc_write(dai->codec, SI476X_DIGITAL_IO_OUTPUT_SAMPLE_RATE,
+ rate);
+ if (err < 0) {
+ dev_err(dai->codec->dev, "Failed to set sample rate\n");
+ return err;
+ }
+
+ err = snd_soc_update_bits(dai->codec, SI476X_DIGITAL_IO_OUTPUT_FORMAT,
+ SI476X_DIGITAL_IO_OUTPUT_WIDTH_MASK,
+ (width << SI476X_DIGITAL_IO_SLOT_SIZE_SHIFT) |
+ (width << SI476X_DIGITAL_IO_SAMPLE_SIZE_SHIFT));
+ if (err < 0) {
+ dev_err(dai->codec->dev, "Failed to set output width\n");
+ return err;
+ }
+
+ return 0;
+}
+
+static int si476x_codec_probe(struct snd_soc_codec *codec)
+{
+ codec->control_data = i2c_mfd_cell_to_core(codec->dev);
+ return 0;
+}
+
+static struct snd_soc_dai_ops si476x_dai_ops = {
+ .hw_params = si476x_codec_hw_params,
+ .set_fmt = si476x_codec_set_dai_fmt,
+};
+
+static struct snd_soc_dai_driver si476x_dai = {
+ .name = "si476x-codec",
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+
+ .rates = SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S8 |
+ SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S20_3LE |
+ SNDRV_PCM_FMTBIT_S24_LE
+ },
+ .ops = &si476x_dai_ops,
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_si476x = {
+ .probe = si476x_codec_probe,
+ .read = si476x_codec_read,
+ .write = si476x_codec_write,
+};
+
+static int si476x_platform_probe(struct platform_device *pdev)
+{
+ return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_si476x,
+ &si476x_dai, 1);
+}
+
+static int si476x_platform_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+ return 0;
+}
+
+MODULE_ALIAS("platform:si476x-codec");
+
+static struct platform_driver si476x_platform_driver = {
+ .driver = {
+ .name = "si476x-codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = si476x_platform_probe,
+ .remove = si476x_platform_remove,
+};
+module_platform_driver(si476x_platform_driver);
+
+MODULE_AUTHOR("Andrey Smirnov <andrey.smirnov@convergeddevices.net>");
+MODULE_DESCRIPTION("ASoC Si4761/64 codec driver");
+MODULE_LICENSE("GPL");
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