IMX6 ALSA question sample rate

clement22 — Fri, 25 Nov 2016 15:46:30 GMT

Hi,

I have a question about how linux deal with audio sample rate.

I'm using an IMX6 SABRE board with an IMX6DL. I replaced the audio codec WM8962 with an other codec.

I implemented a new codec and machine driver and playing/recording works fine.

But something is bugging me :

ALSA always set codec and platform sample rate to 48KHz, no matter what is the actual music sample rate.

For example, here is what I see when I start a music with a sample rate of 176.4KHz :

aplay -v music-wav
M: Rate conversion PCM (48000, sformat=S16_LE)
Converter: linear-interpolation
Protocol version: 10002
Its setup is:
stream       : PLAYBACK
access       : RW_INTERLEAVED
format       : S16_LE
subformat    : STD
channels     : 2
rate         : 176400
exact rate   : 176400 (176400/1)
msbits       : 16
buffer_size : 60211
period_size : 7526
period_time : 42666
tstamp_mode : NONE
tstamp_type : MONOTONIC
period_step : 1
avail_min    : 7526
period_event : 0
start_threshold : 60211
stop_threshold   : 60211
silence_threshold: 0
silence_size : 0
boundary     : 1972994048
Slave: Direct Stream Mixing PCM
Its setup is:
stream       : PLAYBACK
access       : MMAP_INTERLEAVED
format       : S16_LE
subformat    : STD
channels     : 2
rate         : 48000
exact rate   : 48000 (48000/1)
msbits       : 16
buffer_size : 16384
period_size : 2048
period_time : 42666
tstamp_mode : NONE
tstamp_type : MONOTONIC
period_step : 1
avail_min    : 2048
period_event : 0
start_threshold : 16384
stop_threshold   : 16384
silence_threshold: 0
silence_size : 0
boundary     : 1073741824
Hardware PCM card 0 'cs47l24-audio' device 0 subdevice 0
Its setup is:
stream       : PLAYBACK
access       : MMAP_INTERLEAVED
format       : S16_LE
subformat    : STD
channels     : 2
rate         : 48000
exact rate   : 48000 (48000/1)
msbits       : 16
buffer_size : 16384
period_size : 2048
period_time : 42666
tstamp_mode : ENABLE
tstamp_type : MONOTONIC
period_step : 1
avail_min    : 2048
period_event : 0
start_threshold : 1
stop_threshold   : 1073741824
silence_threshold: 0
silence_size : 1073741824
boundary     : 1073741824
appl_ptr     : 0
hw_ptr       : 0

So I'm wondering : why ALSA is configuring codec/platform at 48KHz and resampling the file output instead of configuring directly the codec with the good sample rate ?

In the platform/codec driver there is a structure setting the range of possible sample rate : snd_soc_dai_driver->playback.rates

For codec it's set to : SNDRV_PCM_RATE_8000_192000

For platfrom it's set to : SNDRV_PCM_RATE_CONTINUOUS

It seems to be correctly configured, so why keeping the same sample rate ?

If anybody have an idea, thank you :smileywink:

Re: IMX6 ALSA question sample rate

igorpadykov — Mon, 28 Nov 2016 05:31:36 GMT

Hi Clement

please check imx6qdl.dtsi, in particular fsl,asrc-rate

linux-2.6-imx.git - Freescale i.MX Linux Tree

linux/Documentation/devicetree/bindings/sound/fsl,asrc.txt

and Chapter 29 Asynchronous Sample Rate Converter (ASRC) Driver

attached Linux Manual.

Best regards
igor
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IMX6 ALSA question sample rate

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