From: Lars-Peter Clausen <lars@metafoo.de>
To: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Mike Frysinger <vapier.adi@gmail.com>,
alsa-devel@alsa-project.org, Liam Girdwood <lrg@ti.com>,
device-drivers-devel@blackfin.uclinux.org,
linux-kernel@vger.kernel.org
Subject: Re: [PATCH 2/4] ASoC: Add ADAU1373 codec support
Date: Fri, 12 Aug 2011 02:20:49 +0200 [thread overview]
Message-ID: <4E4471E1.5060706@metafoo.de> (raw)
In-Reply-To: <1313068590.19990.2.camel@finisterre.wolfsonmicro.main>
On 08/11/2011 03:16 PM, Mark Brown wrote:
> On Thu, 2011-08-11 at 12:11 +0200, Lars-Peter Clausen wrote:
>>>> + SOC_ENUM("Lineout1 Mono Stereo", adau1373_lineout1_mode_enum),
>>>> + SOC_ENUM("Speaker Mono Stereo", adau1373_speaker_mode_enum),
>
>>> I'd expect these to be platform data/machine data rather than user
>>> control? The speaker wiring isn't going to vary dynamically...
>
>> You still might want to switch, for whatever particular reason, between mono
>> and stereo at runtime.
>
> Sorry, what does this actually control? I guess I've been mislead but
> from the name of the control I'd expect it to control if the outputs
> were physically connected as mono or stereo outputs.
It controls how Lineout and Speaker mixers distribute the left and right input
channels to their output channels.
>
>>>> + switch (freq / params_rate(params)) {
>>>> + case 1024: /* fs */
>>>> + div = 0;
>>>> + break;
>>>> + case 1536: /* 2/3 fs */
>>>> + div = 1;
>>>> + break;
>
>>> These comments look inaccuate, fs is the sample rate so a divide of 1
>>> would be fs.
>
>> div contains the register value representation of that particular divider.
>
> div isn't the issue here. The comments seem to indicate that the result
> of the divisions are some multiple of fs but fs usually means some
> multiple of the sample rate. For example 2/3fs for 44.1kHz would be
> 29.4kHz.
>
Ah, ok. Yes, I guess the comments are a bit confusing since fs refers to the
internal sample rate and not to the sample rate of the DAI. I'll try to clarify it.
WARNING: multiple messages have this Message-ID (diff)
From: Lars-Peter Clausen <lars@metafoo.de>
To: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Liam Girdwood <lrg@ti.com>,
alsa-devel@alsa-project.org,
device-drivers-devel@blackfin.uclinux.org,
linux-kernel@vger.kernel.org,
Mike Frysinger <vapier.adi@gmail.com>
Subject: Re: [PATCH 2/4] ASoC: Add ADAU1373 codec support
Date: Fri, 12 Aug 2011 02:20:49 +0200 [thread overview]
Message-ID: <4E4471E1.5060706@metafoo.de> (raw)
In-Reply-To: <1313068590.19990.2.camel@finisterre.wolfsonmicro.main>
On 08/11/2011 03:16 PM, Mark Brown wrote:
> On Thu, 2011-08-11 at 12:11 +0200, Lars-Peter Clausen wrote:
>>>> + SOC_ENUM("Lineout1 Mono Stereo", adau1373_lineout1_mode_enum),
>>>> + SOC_ENUM("Speaker Mono Stereo", adau1373_speaker_mode_enum),
>
>>> I'd expect these to be platform data/machine data rather than user
>>> control? The speaker wiring isn't going to vary dynamically...
>
>> You still might want to switch, for whatever particular reason, between mono
>> and stereo at runtime.
>
> Sorry, what does this actually control? I guess I've been mislead but
> from the name of the control I'd expect it to control if the outputs
> were physically connected as mono or stereo outputs.
It controls how Lineout and Speaker mixers distribute the left and right input
channels to their output channels.
>
>>>> + switch (freq / params_rate(params)) {
>>>> + case 1024: /* fs */
>>>> + div = 0;
>>>> + break;
>>>> + case 1536: /* 2/3 fs */
>>>> + div = 1;
>>>> + break;
>
>>> These comments look inaccuate, fs is the sample rate so a divide of 1
>>> would be fs.
>
>> div contains the register value representation of that particular divider.
>
> div isn't the issue here. The comments seem to indicate that the result
> of the divisions are some multiple of fs but fs usually means some
> multiple of the sample rate. For example 2/3fs for 44.1kHz would be
> 29.4kHz.
>
Ah, ok. Yes, I guess the comments are a bit confusing since fs refers to the
internal sample rate and not to the sample rate of the DAI. I'll try to clarify it.
next prev parent reply other threads:[~2011-08-12 0:23 UTC|newest]
Thread overview: 36+ messages / expand[flat|nested] mbox.gz Atom feed top
2011-08-10 3:52 [PATCH 1/4] ASoC: DAPM: Allow multiple mixer sources to be routed via the same switch Lars-Peter Clausen
2011-08-10 3:52 ` Lars-Peter Clausen
2011-08-10 3:52 ` [PATCH 2/4] ASoC: Add ADAU1373 codec support Lars-Peter Clausen
2011-08-10 3:52 ` Lars-Peter Clausen
2011-08-10 6:11 ` Lars-Peter Clausen
2011-08-10 6:11 ` Lars-Peter Clausen
2011-08-11 9:27 ` Mark Brown
2011-08-11 9:27 ` Mark Brown
2011-08-11 10:11 ` Lars-Peter Clausen
2011-08-11 10:11 ` Lars-Peter Clausen
2011-08-11 13:16 ` Mark Brown
2011-08-11 13:16 ` Mark Brown
2011-08-12 0:20 ` Lars-Peter Clausen [this message]
2011-08-12 0:20 ` Lars-Peter Clausen
2011-08-12 1:20 ` Mark Brown
2011-08-12 1:20 ` Mark Brown
2011-08-13 3:25 ` Lars-Peter Clausen
2011-08-13 3:25 ` Lars-Peter Clausen
2011-08-14 10:24 ` Mark Brown
2011-08-14 10:24 ` Mark Brown
2011-08-10 3:52 ` [PATCH 3/4] ASoC: Blackfin: ADAU1373 eval board support Lars-Peter Clausen
2011-08-10 3:52 ` Lars-Peter Clausen
2011-08-10 5:27 ` Scott Jiang
2011-08-10 5:27 ` [alsa-devel] " Scott Jiang
2011-08-10 6:09 ` Lars-Peter Clausen
2011-08-10 6:09 ` [alsa-devel] " Lars-Peter Clausen
2011-08-10 3:52 ` [PATCH 4/4] Blackfin: bf537: Stamp: Register ASoC EVAL-ADAU1373 board driver Lars-Peter Clausen
2011-08-10 3:52 ` Lars-Peter Clausen
2011-08-11 9:31 ` [PATCH 1/4] ASoC: DAPM: Allow multiple mixer sources to be routed via the same switch Mark Brown
2011-08-11 9:31 ` Mark Brown
2011-08-11 9:59 ` Lars-Peter Clausen
2011-08-11 9:59 ` [alsa-devel] " Lars-Peter Clausen
2011-08-12 1:22 ` Mark Brown
-- strict thread matches above, loose matches on Subject: below --
2011-08-15 18:15 Lars-Peter Clausen
2011-08-15 18:15 ` [PATCH 2/4] ASoC: Add ADAU1373 codec support Lars-Peter Clausen
2011-08-16 15:54 ` Mark Brown
2011-08-16 15:54 ` Mark Brown
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