On Wed, Jul 22, 2009 at 05:22:59AM +0200, Janusz Krzysztofik wrote:
Signed-off-by: Janusz Krzysztofik <jkrzyszt@xxxxxxxxxxxx>
---
CPU DAI parameters best matching the codec DAI has been selected out
empirically for best user experience.
Again, all the documentation you've got here could quite happily go in
the commit message and there's a bunch of checkpatch issues.
+#include <sound/soc-dapm.h>
+#include <sound/jack.h>
ASoC will pull that one in for you, not that it really matters.
+static const struct snd_kcontrol_new ams_delta_audio_controls[] = {
+ SOC_ENUM_EXT("Audio Function", ams_delta_audio_enum[0],
+ ams_delta_get_audio_mode, ams_delta_set_audio_mode),
+};
Is it possible to control all the functions of the audio mode
independantly or are only certain combinations possible?
+static struct snd_soc_jack_gpio ams_delta_hook_switch_gpios[];
+static struct snd_soc_jack_pin ams_delta_hook_switch_pins[] = {
+ {
+ .pin = "Mouthpiece",
+ .mask = SND_JACK_MICROPHONE,
+ },
+ {
+ .pin = "Earphone",
+ .mask = SND_JACK_HEADPHONE,
+ },
+ {
+ .pin = "Speaker",
+ .mask = SND_JACK_HEADPHONE,
+ .invert = 1,
+ },
+ {
+ .pin = "Microphone",
+ .mask = SND_JACK_MICROPHONE,
+ .invert = 1,
+ },
+};
I guess microphone and speaker are for speakerphone mode while
mouthpiece and earpiece are a headset? Might be nice to come up with
names that make the paring a bit clearer, or possibly just put a comment
in there.
+/* To actually apply any modem controlled configuration changes to the codec,
+ * we must connect codec DAI pins to the modem for a moment. Be carefull
+ * not to interfere with digital mute function that shares the same hardware. */
+static struct timer_list cx81801_timer;
+static bool cx81801_cmd_pending = 0;
+static bool ams_delta_muted;
+
+static void cx81801_timeout(unsigned long data)
+{
+ /* REVISIT - locking? */
Yeah, locking is probably a good idea :)
+ /* Add board specific DAPM controls */
+ if (!snd_soc_dapm_new_controls(codec, ams_delta_dapm_widgets,
+ ARRAY_SIZE(ams_delta_dapm_widgets))) {
+ if (!snd_soc_dapm_add_routes(codec, ams_delta_audio_map,
+ ARRAY_SIZE(ams_delta_audio_map))) {
The error handling here is a bit odd...
+ /* Add hook switch */
+ if (!snd_soc_jack_new(&ams_delta_audio_card, "hook_switch",
+ SND_JACK_HEADSET, &ams_delta_hook_switch)) {
+ if (!snd_soc_jack_add_gpios(&ams_delta_hook_switch,
+ ARRAY_SIZE(ams_delta_hook_switch_gpios),
+ ams_delta_hook_switch_gpios)) {
+#ifdef CONFIG_GPIO_SYSFS
+ /* Expose hook switch over sysfs if configured */
+ gpio_export(ams_delta_hook_switch_gpios[0].gpio, false);
+#endif
The gpio_export() should be in the ASoC GPIO code rather than here, I'd
expect - care to cook up a patch?