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<p>( 0.090| 0.000) D: [lt-pulseaudio] alsa-mixer.c: Activating
path analog-input-mic<br>
( 0.090| 0.000) D: [lt-pulseaudio] alsa-mixer.c: Path
analog-input-mic (Microphone), direction=2, priority=87,
probed=yes, supported=yes, has_mute=yes, has_volume=yes,
has_dB=yes, min_volume=0, max_volume=48, min_dB=-12, max_dB=12<br>
( 0.090| 0.000) D: [lt-pulseaudio] alsa-mixer.c: Element
Mic, direction=2, switch=1, volume=1, volume_limit=-1,
enumeration=0, required=0, required_any=4, required_absent=0,
mask=0x3600000000f66, n_channels=2, override_map=yes<br>
( 0.090| 0.000) D: [lt-pulseaudio] alsa-mixer.c: Element PCM
Capture Source, direction=2, switch=0, volume=0,
volume_limit=-1, enumeration=1, required=0, required_any=0,
required_absent=0, mask=0x0, n_channels=0, override_map=no<br>
( 0.090| 0.000) D: [lt-pulseaudio] alsa-mixer.c: Option Mic
(analog-input-microphone/analog-input-microphone) index=0,
priority=0<br>
( 0.090| 0.000) D: [lt-pulseaudio] alsa-mixer.c: Element
Line, direction=2, switch=2, volume=2, volume_limit=-1,
enumeration=0, required=0, required_any=0, required_absent=0,
mask=0x6, n_channels=2, override_map=no<br>
( 0.090| 0.000) D: [lt-pulseaudio] alsa-mixer.c: Jack Mic,
alsa_name='Mic Jack', detection unavailable<br>
( 0.090| 0.000) D: [lt-pulseaudio] alsa-mixer.c: Jack Mic
Phantom, alsa_name='Mic Phantom Jack', detection unavailable<br>
( 0.090| 0.000) D: [lt-pulseaudio] alsa-mixer.c: Setting
analog-input-microphone (analog-input-microphone) priority=0<br>
( 0.090| 0.000) D: [lt-pulseaudio] alsa-mixer.c:
element_set_switch(): e->alsa_name Line<br>
( 0.090| 0.000) D: [lt-pulseaudio] alsa-mixer.c:
element_set_switch(): e->direction ==
PA_ALSA_DIRECTION_OUTPUT 33694896</p>
<p>Do you mean the direction is not input ?<br>
</p>
</blockquote>
According to debug message, it is so, however in config:<br>
<blockquote>
<blockquote type="cite">[Mapping analog-stereo-input]<br>
device-strings = hw:%f,1,0<br>
paths-input = analog-input-mic analog-input-linein<br>
channel-map = left,right<br>
direction = input</blockquote>
</blockquote>
There is explicit mentioning of required direction.<br>
<blockquote type="cite">
<p>( 0.090| 0.000) D: [lt-pulseaudio] alsa-mixer.c:
element_set_switch(): b 0</p>
<p>pulseaudio seem just turn off line switch instead of turn on
nic capture volum/switch when activating mic path</p>
<p>One line show element name , values or enum index and enum name
instead of three lines</p>
</blockquote>
Sorry, I'm not sure I understand what do you mean by this.<br>
I'm not C/C++ developer at all, can't even find definition of
snd_mixer_elem_t to find what is that and how to use it (my guess
that you've being asking about it, right?).
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<blockquote type="cite">
<p>( 0.090| 0.000) D: [lt-pulseaudio] alsa-mixer.c:
element_set_option(): e->alsa_name PCM Capture Source<br>
( 0.090| 0.000) D: [lt-pulseaudio] alsa-mixer.c:
element_set_option(): e->switch_use == PA_ALSA_SWITCH_SELECT
0<br>
( 0.090| 0.000) D: [lt-pulseaudio] alsa-mixer.c:
element_set_option(): alsa_idx 0</p>
<p>snd_mixer_selem_get_enum_item_name() provide the name of enum
item by index</p>
</blockquote>
Seems to be the as previous, I really want to help with debugging,
but you should be more concrete since code is too unfamiliar for me.<br>
<blockquote type="cite">
<p>Do you mean you still cannot force pulseaudio to use 24 bits
for your creative usb audio ?</p>
<p>Seem try auto format did not include 24bits </p>
</blockquote>
I have 24bit explicitly in pulseaudio config currently:<br>
<blockquote>
<blockquote type="cite">default-sample-format = s24le</blockquote>
</blockquote>
But it still uses 16bit, didn't managed how to force it to use 24bit<br>
<pre class="moz-signature" cols="72">Sincerely, Nazar Mokrynskyi
github.com/nazar-pc
Skype: nazar-pc
Diaspora: <a class="moz-txt-link-abbreviated" href="mailto:nazarpc@diaspora.mokrynskyi.com">nazarpc@diaspora.mokrynskyi.com</a>
Tox: A9D95C9AA5F7A3ED75D83D0292E22ACE84BA40E912185939414475AF28FD2B2A5C8EF5261249</pre>
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