Hi
I have been trying to make 192/24 files play via HDMI out on the raspberry and I have succeeded to some extend. We can now play files up to 192/24.
I manipulated this file /sound/arm/bcm2835-pcm.c and compiled the kernel and modules
This is part of the original file
And I changed it into:
The problem as I see it is that no matter what file we are playing (44.1/16 or 192/24) it will be played with the correct sample rate that is 44100 and 192000, but all are played with 32 bit depth. So it seems that it only uses the last format like mentioned on the .format line - so now it always use SNDRV_PCM_FMTBIT_S32_LE
All the 24 bit files 88.1 96 qnd 192 plays fine, whereas the lower resolution files 16 bit at 44.1 and 48 kHz plays, but with some distortion.
In order to play all files fine we need to use " -a ::32:0 " for squeezelite and now all files plays fine
If I change this line to: .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE it plays everything at 16 bit
If I change this line to .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE it only produces noise
If I change this line to .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_3LE it only produces noise
So there seems to be a couple of questions:
1. why does it only use the the last option in the .formats line?
2. why can't it use the 24 bits format?
3. The change from 16 and 24 bit to 32 bit when playing will it negatively affect the sound quality. I cant hear any difference, but my equipment is not so good.
Steen
Here is the output when playing a 44.1/16 flac file
cat /proc/asound/card0/pcm0p/sub0/hw_params
access: RW_INTERLEAVED
format: S32_LE
subformat: STD
channels: 2
rate: 44100 (44100/1)
period_size: 442
buffer_size: 1768
here when 192/24 bit flac is played:
cat /proc/asound/card0/pcm0p/sub0/hw_params
access: RW_INTERLEAVED
format: S32_LE
subformat: STD
channels: 2
rate: 192000 (192000/1)
period_size: 1920
buffer_size: 7680
I have been trying to make 192/24 files play via HDMI out on the raspberry and I have succeeded to some extend. We can now play files up to 192/24.
I manipulated this file /sound/arm/bcm2835-pcm.c and compiled the kernel and modules
This is part of the original file
Code:
<linux/interrupt.h>
#include <linux/slab.h>
#include "bcm2835.h"
/* hardware definition */
static struct snd_pcm_hardware snd_bcm2835_playback_hw = {
.info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID),
.formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
.rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
.rate_min = 8000,
.rate_max = 48000,
.channels_min = 1,
.channels_max = 2,
.buffer_bytes_max = 128 * 1024,
.period_bytes_min = 1 * 1024,
.period_bytes_max = 128 * 1024,
.periods_min = 1,
.periods_max = 128,
};Code:
#include <linux/slab.h>
#include "bcm2835.h"
/* hardware definition */
static struct snd_pcm_hardware snd_bcm2835_playback_hw = {
.info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID),
.formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE,
.rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_192000,
.rate_min = 8000,
.rate_max = 1920000,
.channels_min = 1,
.channels_max = 2,
.buffer_bytes_max = 256 * 1024,
.period_bytes_min = 1 * 1024,
.period_bytes_max = 256 * 1024,
.periods_min = 1,
.periods_max = 256,
};The problem as I see it is that no matter what file we are playing (44.1/16 or 192/24) it will be played with the correct sample rate that is 44100 and 192000, but all are played with 32 bit depth. So it seems that it only uses the last format like mentioned on the .format line - so now it always use SNDRV_PCM_FMTBIT_S32_LE
All the 24 bit files 88.1 96 qnd 192 plays fine, whereas the lower resolution files 16 bit at 44.1 and 48 kHz plays, but with some distortion.
In order to play all files fine we need to use " -a ::32:0 " for squeezelite and now all files plays fine
If I change this line to: .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE it plays everything at 16 bit
If I change this line to .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE it only produces noise
If I change this line to .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_3LE it only produces noise
So there seems to be a couple of questions:
1. why does it only use the the last option in the .formats line?
2. why can't it use the 24 bits format?
3. The change from 16 and 24 bit to 32 bit when playing will it negatively affect the sound quality. I cant hear any difference, but my equipment is not so good.
Steen
Here is the output when playing a 44.1/16 flac file
cat /proc/asound/card0/pcm0p/sub0/hw_params
access: RW_INTERLEAVED
format: S32_LE
subformat: STD
channels: 2
rate: 44100 (44100/1)
period_size: 442
buffer_size: 1768
here when 192/24 bit flac is played:
cat /proc/asound/card0/pcm0p/sub0/hw_params
access: RW_INTERLEAVED
format: S32_LE
subformat: STD
channels: 2
rate: 192000 (192000/1)
period_size: 1920
buffer_size: 7680