Thank you Glenn!
On Wednesday, 30 January 2013 03:14:13 UTC+1, Glenn Kasten wrote:
>
> 1. It's code in the HAL. The HAL code figures out what the use case is,
> constructs the corresponding mixer path string, then uses mixer_paths.xml
> to apply the right settings corresponding to that path.
> See also audio_policy.conf, but it's not directly related to this.
>
> 2. I'm not aware of such a restriction.
>
> 3. That's the hardware volume, which is mostly unused currently.
> AudioFlinger applies a software attenuation instead.
>
> On Tuesday, January 29, 2013 1:26:40 AM UTC-8, ffxx68 wrote:
>>
>> I think I'm starting to figuring out what to fix. My ALC5621 kernel
>> driver includes these control definitions:
>>
>> static struct snd_kcontrol_new emxx_codec_controls[] = {
>> EMXX_CODEC_INTEGER("DAC Volume", 0, MIXER_VOL_DAC),
>> EMXX_CODEC_INTEGER("Headphone Volume", 0, MIXER_VOL_HP),
>> EMXX_CODEC_INTEGER("Speaker Volume", 0, MIXER_VOL_SPK),
>> EMXX_CODEC_INTEGER("MIC1 Volume", 0, MIXER_VOL_MIC1),
>> EMXX_CODEC_INTEGER("MIC2 Volume", 0, MIXER_VOL_MIC2),
>> EMXX_CODEC_INTEGER("AUXIN Volume", 0, MIXER_VOL_AUXIN),
>> EMXX_CODEC_INTEGER("AUXOUT Volume", 0, MIXER_VOL_AUXOUT),
>> EMXX_CODEC_ENUM("Capture Switch", 0, MIXER_SW_CAPTURE),
>> EMXX_CODEC_ENUM("Sampling Rate Switch", 0, MIXER_SW_SAMPLING_RATE),
>> EMXX_CODEC_ENUM("Playback Switch", 0, MIXER_SW_PLAYBACK),
>> EMXX_CODEC_BOOLEAN("CODEC Power Switch", 0, MIXER_SW_CODEC_POWER_BL),
>> };
>>
>> ("snd_kcontrol_new" is a standard structure, used by grouper rt5640 as
>> well).
>>
>> Values are selected from these sets, respectively:
>>
>> "Playback Switch" -> "OFF", "Speaker_normal", "Speaker_ringtone",
>> "Speaker_incall", "Earpiece_ringtone", "Earpiece_incall", "Headset_normal",
>> "Headset_ringtone", "Headset_incall"
>> "Capture Switch" -> "OFF", "MIC_normal", "MIC_ringtone", "MIC_incall",
>> "Headset_normal", "Headset_ringtone", "Headset_incall"
>> "Sampling Rate Switch" -> "7.35kHz", "8kHz", "11.025kHz", "12kHz",
>> "14.7kHz", "16kHz", "22.05kHz", "24kHz", "29.4kHz", "32kHz", "44.1kHz",
>> "48kHz"
>> "CODEC Power Switch" -> 0, 1
>>
>> I'm now going to map these in mixer_paths.xml.
>>
>> A few question though:
>>
>> 1) how is a given routing selected, out of those defined in
>> mixer_paths.xml?
>> 2) Is it true (as I read somewhere) that tinyalsa supports a 44.1KHz
>> sampling rate only?
>> 3) why in grouper hw_audio.c the volume-setting functions are empty?
>> How's volume set?
>>
>> static int out_set_volume(struct audio_stream_out *stream, float left,
>> float right)
>> {
>> return -ENOSYS;
>> }
>>
>>
>>
>>
>>
>>
>>
>> On Monday, 28 January 2013 17:23:33 UTC+1, ffxx68 wrote:
>>>
>>> Hi Glenn, thanks a lot for answering.
>>>
>>> I hope you don't mind if I ask you to guide me through these steps. I
>>> have both the kernel driver code, the datasheet and device schematics, so I
>>> have all's required to complete this... I think. Except experience :)
>>>
>>> For example, I've posted the device control names already, from kernel
>>> driver.
>>> There's no external DSP, as the audio section is relatively simple and
>>> designed around a Realtek ALC5621 chip.
>>> About control values. I've got the ALC5621 datasheet, but I miss where
>>> to define those values in our code. If you could provide how that works for
>>> the grouper' example, for just one control, I could try to map to my device
>>> and eventually extend to all controls.
>>>
>>> On Thursday, 8 November 2012 14:42:21 UTC+1, ffxx68 wrote:
>>>>
>>>> Hi
>>>>
>>>> I'm trying to port a ICS implementation of ALSA audio to JB, or a
>>>> tablet device, under this project:
>>>>
>>>> Mainly, I have the following to integrate (which I have the patch for
>>>> ICS):
>>>>
>>>> external/alsa-lib
>>>> hardware/alsa_sound
>>>>
>>>> plus some other smaller fix around, but I want to understand first of
>>>> all what to do with these two first.
>>>>
>>>> For example, I have the some files with the same names in both
>>>> hardware/alsa_sound and hardware/libhardware_legacy/
>>>> audio (from AOSP), but their content is very different in the two
>>>> locations.
>>>>
>>>> If I keep the ones from hardware/alsa_sound, build fails.
>>>> If I copy them from hardware/libhardware_legacy/audio to
>>>> hardware/alsa_sound, I can build but I get a crash during boot.
>>>>
>>>> Which one should I keep and compile?
>>>> Once compiled, which libraries should I use from alsa_sound or
>>>> libhardware_legacy/audio?
>>>>
>>>> Basically, I don't know the approach to follow, with the ALSA
>>>> integration. I couldn't find any guide, or tutorial, so any help in that
>>>> sense is welcome too.
>>>>
>>>> Thanks in advance
>>>> Fabio
>>>>
>>>
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