rockbox/firmware/drivers/audio/nwzlinux-codec.c

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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
*
* Copyright (c) 2016 Amaury Pouly
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
#include "logf.h"
#include "system.h"
#include "kernel.h"
#include "string.h"
#include "stdio.h"
#include "audio.h"
#include "sound.h"
#include "audiohw.h"
#include "cscodec.h"
#include "nwzlinux_codec.h"
#include "stdlib.h"
#include "panic.h"
#include <sys/ioctl.h>
#include "nwz_audio.h"
#include "pcm-alsa.h"
#include "alsa-controls.h"
/* This driver handle the Sony linux audio drivers: despite using many differents
* codecs, it appears that they all share a common interface and common controls. */
/* This is the alsa mixer interface exposed by Sony:
numid=3,iface=MIXER,name='Capture Src Switch'
; type=ENUMERATED,access=rw------,values=1,items=4
; Item #0 'None'
; Item #1 'Line'
; Item #2 'Fm'
; Item #3 'Mic'
: values=0
numid=2,iface=MIXER,name='Playback Src Switch'
; type=ENUMERATED,access=rw------,values=1,items=7
; Item #0 'None'
; Item #1 'Music'
; Item #2 'Video'
; Item #3 'Tv'
; Item #4 'Fm'
; Item #5 'Line'
; Item #6 'Mic'
: values=1
numid=1,iface=MIXER,name='Playback Volume'
; type=INTEGER,access=rw------,values=2,min=0,max=100,step=1
: values=5,5
numid=7,iface=MIXER,name='CODEC Acoustic Switch'
; type=BOOLEAN,access=rw------,values=1
: values=on
numid=8,iface=MIXER,name='CODEC Cue/Rev Switch'
; type=BOOLEAN,access=rw------,values=1
: values=off
numid=9,iface=MIXER,name='CODEC Fade In Switch'
; type=BOOLEAN,access=rw------,values=1
: values=off
numid=6,iface=MIXER,name='CODEC Mute Switch'
; type=BOOLEAN,access=rw------,values=1
: values=off
numid=5,iface=MIXER,name='CODEC Power Switch'
; type=BOOLEAN,access=rw------,values=1
: values=on
numid=10,iface=MIXER,name='CODEC Stanby Switch'
; type=BOOLEAN,access=rw------,values=1
: values=off
numid=4,iface=MIXER,name='Output Switch'
; type=ENUMERATED,access=rw------,values=1,items=4
; Item #0 'Headphone'
; Item #1 'LineVariable'
; Item #2 'LineFixed'
; Item #3 'Speaker'
: values=0
numid=6,iface=MIXER,name='Sampling Rate'
; type=ENUMERATED,access=rw------,values=1,items=6
; Item #0 '44100'
; Item #1 '48000'
; Item #2 '88200'
; Item #3 '96000'
; Item #4 '176400'
; Item #5 '192000'
: values=0
*/
/* List of various codecs used by Sony */
enum nwz_codec_t
{
NWZ_CS42L56,
NWZ_R2A15602LG_D,
NWZ_CS47L01_A,
NWZ_CS47L01_D,
NWZ_CXD3774GF_D,
NWZ_UNK_CODEC,
};
#define NWZ_LEVEL_MUTE -1000
/* Description of the volume curve implemented by the kernel driver */
struct nwz_vol_curve_t
{
int count; /* number of levels */
int level[]; /* levels in tenth-dB, level[0] is always mute */
};
/* file descriptor of the icx_noican device */
static int fd_noican;
/* file descriptor of the hardware sound device */
static int fd_hw;
/* Codec */
static enum nwz_codec_t nwz_codec;
/* does the code support setting the sample rate? */
static bool has_sample_rate;
static enum nwz_codec_t find_codec(void)
{
if(nwz_is_kernel_module_loaded("cs42L56_d"))
return NWZ_CS42L56;
if(nwz_is_kernel_module_loaded("r2A15602LG_d"))
return NWZ_R2A15602LG_D;
if(nwz_is_kernel_module_loaded("cs47L01_d"))
return NWZ_CS47L01_D;
if(nwz_is_kernel_module_loaded("cs47L01_a"))
return NWZ_CS47L01_A;
if(nwz_is_kernel_module_loaded("cxd3774gf_d"))
return NWZ_CXD3774GF_D;
return NWZ_UNK_CODEC;
}
const char *nwz_get_codec_name(void)
{
switch(nwz_codec)
{
case NWZ_CS42L56: return "cs42L56_d";
case NWZ_R2A15602LG_D: return "r2A15602LG_d";
case NWZ_CS47L01_D: return "cs47L01_d";
case NWZ_CS47L01_A: return "cs47L01_a";
case NWZ_CXD3774GF_D: return "cxd3774gf_d";
default: return "Unknown";
}
}
static struct nwz_vol_curve_t cxd3774gf_vol_curve =
{
.count = 31,
/* Most Sonys seem to follow the convention of 3dB/step then 2dB/step then 1dB/step */
.level = {NWZ_LEVEL_MUTE,
-550, -520, -490, -460, -430, -400, -370, -340, -310, -280, -250, /* 3dB/step */
-230, -210, -190, -170, -150, -130, -110, -90, /* 2dB/step */
-80, -70, -60, -50, -40, -30, -20, -10, 0, /* 1dB/step */
15, 35, /* 1.5dB then 2dB */
}
};
struct nwz_vol_curve_t *nwz_get_codec_vol_curve(void)
{
switch(nwz_codec)
{
case NWZ_CS47L01_A:
case NWZ_CS47L01_D:
/* there are 32 levels but the last two are the same so in fact it
* is the same curve as the cxd3774gf_d */
case NWZ_CXD3774GF_D:
return &cxd3774gf_vol_curve;
default:
/* return the safest curve (only 31 levels) */
return &cxd3774gf_vol_curve;
}
}
static void noican_init(void)
{
fd_noican = open(NWZ_NC_DEV, O_RDWR);
/* some targets don't have noise cancelling so silently fail */
}
static void noican_close(void)
{
if(fd_noican >= 0)
close(fd_noican);
}
/* Set NC switch */
static void noican_set_switch(int sw)
{
if(ioctl(fd_noican, NWZ_NC_SET_SWITCH, &sw) < 0)
panicf("ioctl(NWZ_NC_SET_SWITCH) failed");
}
/* Get NC switch */
static int noican_get_switch(void)
{
int val;
if(ioctl(fd_noican, NWZ_NC_GET_SWITCH, &val) < 0)
panicf("ioctl(NWZ_NC_GET_SWITCH) failed");
return val;
}
/* Get HP status */
static int noican_get_hp_status(void)
{
int val;
if(ioctl(fd_noican, NWZ_NC_GET_HP_STATUS, &val) < 0)
panicf("ioctl(NWZ_NC_GET_HP_STATUS) failed");
return val;
}
/* Set HP type */
static void noican_set_hp_type(int type)
{
if(ioctl(fd_noican, NWZ_NC_SET_HP_TYPE, &type) < 0)
panicf("ioctl(NWZ_NC_SET_HP_TYPE) failed");
}
/* Get HP type */
static int noican_get_hp_type(void)
{
int val;
if(ioctl(fd_noican, NWZ_NC_GET_HP_TYPE, &val) < 0)
panicf("ioctl(NWZ_NC_GET_HP_TYPE) failed");
return val;
}
/* Set gain */
static void noican_set_gain(int gain)
{
if(ioctl(fd_noican, NWZ_NC_SET_GAIN, &gain) < 0)
panicf("ioctl(NWZ_NC_SET_GAIN) failed");
}
/* Get gain */
static int noican_get_gain(void)
{
int val;
if(ioctl(fd_noican, NWZ_NC_GET_GAIN, &val) < 0)
panicf("ioctl(NWZ_NC_GET_GAIN) failed");
return val;
}
/* Set filter */
static void noican_set_filter(int filter)
{
if(ioctl(fd_noican, NWZ_NC_SET_FILTER, &filter) < 0)
panicf("ioctl(NWZ_NC_SET_FILTER) failed");
}
/* Get filter */
static int noican_get_filter(void)
{
int val;
if(ioctl(fd_noican, NWZ_NC_GET_FILTER, &val) < 0)
panicf("ioctl(NWZ_NC_GET_FILTER) failed");
return val;
}
static void hw_open(void)
{
fd_hw = open("/dev/snd/hwC0D0", O_RDWR);
if(fd_hw < 0)
panicf("Cannot open '/dev/snd/hwC0D0'");
}
static void hw_close(void)
{
close(fd_hw);
}
void audiohw_preinit(void)
{
alsa_controls_init();
/* turn on codec */
alsa_controls_set_bool("CODEC Power Switch", true);
/* mute */
alsa_controls_set_bool("CODEC Mute Switch", true);
/* Acoustic and Cue/Rev control how the volume curve, but it is not clear
* what the intention of these modes are and the OF does not seem to use
* them by default */
alsa_controls_set_bool("CODEC Acoustic Switch", false);
alsa_controls_set_bool("CODEC Cue/Rev Switch", false);
/* not sure exactly what it means */
alsa_controls_set_enum("Playback Src Switch", "Music");
/* use headphone output */
alsa_controls_set_enum("Output Switch", "Headphone");
/* unmute */
alsa_controls_set_bool("CODEC Mute Switch", false);
/* sample rate */
has_sample_rate = alsa_has_control("Sampling Rate");
/* init noican */
noican_init();
if(fd_noican >= 0)
{
/* dump configuration, for debug purposes */
printf("nc hp status: %d\n", noican_get_hp_status());
printf("nc type: %d\n", noican_get_hp_type());
printf("nc switch: %d\n", noican_get_switch());
printf("nc gain: %d\n", noican_get_gain());
printf("nc filter: %d\n", noican_get_filter());
/* make sure we start in a clean state */
noican_set_switch(NWZ_NC_SWITCH_OFF);
noican_set_hp_type(NC_HP_TYPE_DEFAULT);
noican_set_filter(NWZ_NC_FILTER_INDEX_0);
noican_set_gain(NWZ_NC_GAIN_CENTER);
}
/* init hw */
hw_open();
nwz_codec = find_codec();
printf("Codec: %s\n", nwz_get_codec_name());
}
void audiohw_postinit(void)
{
}
/* volume must be driver unit */
static void nwz_set_driver_vol(int vol)
{
long vols[2];
/* the driver expects percent, convert from centibel in range 0...x */
vols[0] = vols[1] = vol;
/* on some recent players like A10, Sony decided to merge left/right volume
* into one, thus we need to make sure we write the correct number of values */
int vol_cnt;
alsa_controls_get_info("Playback Volume", &vol_cnt);
alsa_controls_set_ints("Playback Volume", vol_cnt, vols);
}
/* volume is in tenth-dB */
void audiohw_set_volume(int vol_l, int vol_r)
{
/* FIXME at the moment we don't support balance and just average left and right.
* But this could be implemented using pcm alsa digital volume */
/* the Sony drivers expect vol_l = vol_r */
int vol = (vol_l + vol_r) / 2;
printf("request volume %d dB\n", vol / 10);
struct nwz_vol_curve_t *curve = nwz_get_codec_vol_curve();
/* min/max for pcm volume */
int min_pcm = -430;
int max_pcm = 0;
/* On some codecs (like cs47L01), Sony clear overdrives the DAC which produces
* massive clipping at any level (since they fix the DAC volume at around +6dB
* and then adjust HP volume in negative at the top of range !!). The only
* solution around this problem is to use the digital volume first so that
* very quickly the digital volume compensate for the DAC overdrive and we
* avoid clipping. */
int sony_clip_level = -80; /* any volume above this will cause massive clipping the DAC */
/* to avoid the clipping problem, virtually decrease requested volume by the
* clipping threshold, so that we will compensate in digital later by
* at least this amount if possibly */
vol -= sony_clip_level;
int drv_vol = curve->count - 1;
/* pick driver level just above request volume */
while(drv_vol > 0 && curve->level[drv_vol - 1] >= vol)
drv_vol--;
/* now remove the artifical volume change */
vol += sony_clip_level;
/* now adjust digital volume */
vol -= curve->level[drv_vol];
if(vol < min_pcm)
{
vol = min_pcm; /* digital cannot do <43dB */
drv_vol = 0; /* mute */
}
else if(vol > max_pcm)
vol = max_pcm; /* digital cannot do >0dB */
printf(" set driver volume %d (%d dB)\n", drv_vol, curve->level[drv_vol] / 10);
nwz_set_driver_vol(drv_vol);
printf(" set digital volume %d dB\n", vol / 10);
pcm_alsa_set_digital_volume(vol / 10);
}
void audiohw_close(void)
{
hw_close();
alsa_controls_close();
noican_close();
}
void audiohw_set_frequency(int fsel)
{
if(has_sample_rate)
{
/* it's slightly annoying that Sony put the value in an enum with strings... */
char freq_str[16];
sprintf(freq_str, "%d", fsel);
alsa_controls_set_enum("Sampling Rate", freq_str);
}
}