[alsa-devel] Questions about Multi-streaming playback with HDA codec ad1988b
Raymond Yau
superquad.vortex2 at gmail.com
Thu Jan 13 03:09:43 CET 2011
The patch is to provide a new model which front panel headphone of a
6stack-dig desktop can be used for playback a stereo stream independent of
the rear panel
ad1988b is a HDA codec support 10 channel ( enough DAC for stereo +
surround71 )
1) Is this approach to create a new model correct or just need to implement
a switch for activatie the "independent head phone" like those VIA codecs ?
struct ad198x_spec {
+ hda_nid_t *alt_dac_nid;
+ struct hda_pcm_stream *stream_analog_alt_playback;
}
+static hda_nid_t ad1988_6stack_alt_dac_nid[1] = {
+ 0x03
+};
static const char *ad1988_models[AD1988_MODEL_LAST] = {
[AD1988_6STACK] = "6stack",
[AD1988_6STACK_DIG] = "6stack-dig",
+ [AD1988_6STACK_DIG_FP] = "6stack-dig-fp",
[AD1988_3STACK] = "3stack",
[AD1988_3STACK_DIG] = "3stack-dig",
[AD1988_LAPTOP] = "laptop",
[AD1988_LAPTOP_DIG] = "laptop-dig",
[AD1988_AUTO] = "auto",
};
enum {
AD1988_6STACK,
AD1988_6STACK_DIG,
+ AD1988_6STACK_DIG_FP,
AD1988_3STACK,
AD1988_3STACK_DIG,
AD1988_LAPTOP,
AD1988_LAPTOP_DIG,
AD1988_AUTO,
AD1988_MODEL_LAST,
};
2) do the driver need to define .prepare and .cleanup function or just need
to use "hda_pcm_default_prepare() and hda_pcm_default_cleanup()
static int ad198x_alt_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
unsigned int stream_tag,
unsigned int format,
struct snd_pcm_substream *substream)
{
struct ad198x_spec *spec = codec->spec;
snd_hda_codec_setup_stream(codec, spec->alt_dac_nid[0], stream_tag, 0,
format);
return 0;
}
static int ad198x_alt_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
struct ad198x_spec *spec = codec->spec;
snd_hda_codec_cleanup_stream(codec, spec->alt_dac_nid[0]);
return 0;
}
static struct hda_pcm_stream ad198x_pcm_analog_alt_playback = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
/* NID is set in ad198x_build_pcms */
.ops = {
.prepare = ad198x_alt_playback_pcm_prepare,
.cleanup = ad198x_alt_playback_pcm_cleanup
},
};
3) Is it correct to use device 2 for analog alternate playback since some
HDA codec implement this feature on "hw:0,0,1" and realtek driver create
alternate capture on "hw:0,2" , this seem create a lot of problem of the
application (e.g. PA server ) ?
static int ad198x_build_pcms(struct hda_codec *codec)
{
if (spec->alt_dac_nid && spec->stream_analog_alt_playback) {
codec->num_pcms++;
info = spec->pcm_rec + 2;
info->name = "AD198x Alternate Analog";
info->pcm_type = HDA_PCM_TYPE_AUDIO;
info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
*spec->stream_analog_alt_playback;
info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->alt_dac_nid[0];
}
}
static struct snd_kcontrol_new ad1988_6stack_fp_mixers[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x06, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0,
HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Side Playback Volume", 0x0a, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x2a, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x27, 1, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x27, 2, 2, HDA_INPUT),
HDA_BIND_MUTE("Side Playback Switch", 0x28, 2, HDA_INPUT),
HDA_BIND_MUTE("Headphone Playback Switch", 0x22, 2, HDA_INPUT),
HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT),
HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT),
{ } /* end */
};
static struct hda_verb ad1988_6stack_fp_init_verbs[] = {
/* Front, Surround, CLFE, side DAC; unmute as default */
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Headphone; unmute as default */
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Port-A front headphon path */
{0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC:03h */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
/* Port-D line-out path */
{0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* Port-F surround path */
{0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* Port-G CLFE path */
{0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* Port-H side path */
{0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* Mono out path */
{0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */
{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */
/* Port-B front mic-in path */
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* Port-C line-in path */
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x33, AC_VERB_SET_CONNECT_SEL, 0x0},
/* Port-E mic-in path */
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x34, AC_VERB_SET_CONNECT_SEL, 0x0},
/* Analog CD Input */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* Analog Mix output amp */
{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
{ }
};
static int patch_ad1988(struct hda_codec *codec)
{
switch (board_config) {
case AD1988_6STACK_DIG_FP:
spec->multiout.max_channels = 8;
spec->multiout.num_dacs = 4;
spec->multiout.dac_nids = ad1988_6stack_dac_nids;
spec->alt_dac_nid = ad1988_6stack_alt_dac_nid;
spec->stream_analog_alt_playback = &ad198x_pcm_analog_alt_playback;
spec->input_mux = &ad1988_6stack_capture_source;
spec->num_mixers = 1;
spec->mixers[0] = ad1988_6stack_fp_mixers;
spec->num_init_verbs = 1;
spec->init_verbs[0] = ad1988_6stack_fp_init_verbs;
spec->multiout.dig_out_nid = AD1988_SPDIF_OUT;
spec->dig_in_nid = AD1988_SPDIF_IN;
break;
3) What functions are still missing in this model ?
a), do this new model need to disable hp_automute() function
b) any change in ad198x_power_eapd_xxx() functions ?
c) idenifty ad1988a and ad1988b
More information about the Alsa-devel
mailing list