afbae177a1
We don't really know what those are supposed to do. They seem to change the volume curve but it is not very clear what is the intended purpose. Change-Id: I65f5d18aba139844c23df092277ba17ee8518f96
427 lines
12 KiB
C
427 lines
12 KiB
C
/***************************************************************************
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* __________ __ ___.
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* Open \______ \ ____ ____ | | _\_ |__ _______ ___
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* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
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* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
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* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
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* \/ \/ \/ \/ \/
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*
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* Copyright (c) 2016 Amaury Pouly
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* as published by the Free Software Foundation; either version 2
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* of the License, or (at your option) any later version.
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*
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* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
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* KIND, either express or implied.
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*
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****************************************************************************/
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#include "logf.h"
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#include "system.h"
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#include "kernel.h"
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#include "string.h"
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#include "stdio.h"
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#include "audio.h"
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#include "sound.h"
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#include "audiohw.h"
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#include "cscodec.h"
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#include "nwzlinux_codec.h"
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#include "stdlib.h"
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#include "panic.h"
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#include <sys/ioctl.h>
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#include "nwz_audio.h"
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#include "pcm-alsa.h"
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#include "alsa-controls.h"
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/* This driver handle the Sony linux audio drivers: despite using many differents
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* codecs, it appears that they all share a common interface and common controls. */
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/* This is the alsa mixer interface exposed by Sony:
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numid=3,iface=MIXER,name='Capture Src Switch'
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; type=ENUMERATED,access=rw------,values=1,items=4
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; Item #0 'None'
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; Item #1 'Line'
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; Item #2 'Fm'
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; Item #3 'Mic'
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: values=0
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numid=2,iface=MIXER,name='Playback Src Switch'
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; type=ENUMERATED,access=rw------,values=1,items=7
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; Item #0 'None'
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; Item #1 'Music'
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; Item #2 'Video'
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; Item #3 'Tv'
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; Item #4 'Fm'
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; Item #5 'Line'
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; Item #6 'Mic'
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: values=1
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numid=1,iface=MIXER,name='Playback Volume'
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; type=INTEGER,access=rw------,values=2,min=0,max=100,step=1
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: values=5,5
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numid=7,iface=MIXER,name='CODEC Acoustic Switch'
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; type=BOOLEAN,access=rw------,values=1
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: values=on
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numid=8,iface=MIXER,name='CODEC Cue/Rev Switch'
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; type=BOOLEAN,access=rw------,values=1
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: values=off
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numid=9,iface=MIXER,name='CODEC Fade In Switch'
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; type=BOOLEAN,access=rw------,values=1
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: values=off
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numid=6,iface=MIXER,name='CODEC Mute Switch'
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; type=BOOLEAN,access=rw------,values=1
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: values=off
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numid=5,iface=MIXER,name='CODEC Power Switch'
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; type=BOOLEAN,access=rw------,values=1
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: values=on
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numid=10,iface=MIXER,name='CODEC Stanby Switch'
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; type=BOOLEAN,access=rw------,values=1
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: values=off
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numid=4,iface=MIXER,name='Output Switch'
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; type=ENUMERATED,access=rw------,values=1,items=4
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; Item #0 'Headphone'
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; Item #1 'LineVariable'
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; Item #2 'LineFixed'
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; Item #3 'Speaker'
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: values=0
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numid=6,iface=MIXER,name='Sampling Rate'
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; type=ENUMERATED,access=rw------,values=1,items=6
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; Item #0 '44100'
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; Item #1 '48000'
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; Item #2 '88200'
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; Item #3 '96000'
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; Item #4 '176400'
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; Item #5 '192000'
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: values=0
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*/
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/* List of various codecs used by Sony */
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enum nwz_codec_t
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{
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NWZ_CS42L56,
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NWZ_R2A15602LG_D,
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NWZ_CS47L01_A,
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NWZ_CS47L01_D,
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NWZ_CXD3774GF_D,
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NWZ_UNK_CODEC,
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};
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#define NWZ_LEVEL_MUTE -1000
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/* Description of the volume curve implemented by the kernel driver */
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struct nwz_vol_curve_t
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{
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int count; /* number of levels */
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int level[]; /* levels in tenth-dB, level[0] is always mute */
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};
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/* file descriptor of the icx_noican device */
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static int fd_noican;
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/* file descriptor of the hardware sound device */
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static int fd_hw;
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/* Codec */
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static enum nwz_codec_t nwz_codec;
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/* does the code support setting the sample rate? */
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static bool has_sample_rate;
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static enum nwz_codec_t find_codec(void)
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{
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if(nwz_is_kernel_module_loaded("cs42L56_d"))
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return NWZ_CS42L56;
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if(nwz_is_kernel_module_loaded("r2A15602LG_d"))
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return NWZ_R2A15602LG_D;
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if(nwz_is_kernel_module_loaded("cs47L01_d"))
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return NWZ_CS47L01_D;
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if(nwz_is_kernel_module_loaded("cs47L01_a"))
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return NWZ_CS47L01_A;
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if(nwz_is_kernel_module_loaded("cxd3774gf_d"))
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return NWZ_CXD3774GF_D;
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return NWZ_UNK_CODEC;
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}
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const char *nwz_get_codec_name(void)
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{
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switch(nwz_codec)
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{
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case NWZ_CS42L56: return "cs42L56_d";
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case NWZ_R2A15602LG_D: return "r2A15602LG_d";
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case NWZ_CS47L01_D: return "cs47L01_d";
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case NWZ_CS47L01_A: return "cs47L01_a";
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case NWZ_CXD3774GF_D: return "cxd3774gf_d";
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default: return "Unknown";
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}
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}
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static struct nwz_vol_curve_t cxd3774gf_vol_curve =
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{
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.count = 31,
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/* Most Sonys seem to follow the convention of 3dB/step then 2dB/step then 1dB/step */
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.level = {NWZ_LEVEL_MUTE,
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-550, -520, -490, -460, -430, -400, -370, -340, -310, -280, -250, /* 3dB/step */
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-230, -210, -190, -170, -150, -130, -110, -90, /* 2dB/step */
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-80, -70, -60, -50, -40, -30, -20, -10, 0, /* 1dB/step */
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15, 35, /* 1.5dB then 2dB */
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}
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};
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struct nwz_vol_curve_t *nwz_get_codec_vol_curve(void)
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{
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switch(nwz_codec)
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{
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case NWZ_CS47L01_A:
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case NWZ_CS47L01_D:
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/* there are 32 levels but the last two are the same so in fact it
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* is the same curve as the cxd3774gf_d */
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case NWZ_CXD3774GF_D:
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return &cxd3774gf_vol_curve;
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default:
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/* return the safest curve (only 31 levels) */
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return &cxd3774gf_vol_curve;
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}
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}
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static void noican_init(void)
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{
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fd_noican = open(NWZ_NC_DEV, O_RDWR);
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/* some targets don't have noise cancelling so silently fail */
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}
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static void noican_close(void)
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{
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if(fd_noican >= 0)
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close(fd_noican);
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}
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/* Set NC switch */
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static void noican_set_switch(int sw)
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{
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if(ioctl(fd_noican, NWZ_NC_SET_SWITCH, &sw) < 0)
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panicf("ioctl(NWZ_NC_SET_SWITCH) failed");
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}
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/* Get NC switch */
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static int noican_get_switch(void)
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{
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int val;
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if(ioctl(fd_noican, NWZ_NC_GET_SWITCH, &val) < 0)
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panicf("ioctl(NWZ_NC_GET_SWITCH) failed");
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return val;
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}
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/* Get HP status */
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static int noican_get_hp_status(void)
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{
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int val;
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if(ioctl(fd_noican, NWZ_NC_GET_HP_STATUS, &val) < 0)
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panicf("ioctl(NWZ_NC_GET_HP_STATUS) failed");
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return val;
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}
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/* Set HP type */
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static void noican_set_hp_type(int type)
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{
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if(ioctl(fd_noican, NWZ_NC_SET_HP_TYPE, &type) < 0)
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panicf("ioctl(NWZ_NC_SET_HP_TYPE) failed");
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}
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/* Get HP type */
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static int noican_get_hp_type(void)
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{
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int val;
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if(ioctl(fd_noican, NWZ_NC_GET_HP_TYPE, &val) < 0)
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panicf("ioctl(NWZ_NC_GET_HP_TYPE) failed");
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return val;
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}
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/* Set gain */
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static void noican_set_gain(int gain)
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{
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if(ioctl(fd_noican, NWZ_NC_SET_GAIN, &gain) < 0)
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panicf("ioctl(NWZ_NC_SET_GAIN) failed");
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}
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/* Get gain */
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static int noican_get_gain(void)
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{
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int val;
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if(ioctl(fd_noican, NWZ_NC_GET_GAIN, &val) < 0)
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panicf("ioctl(NWZ_NC_GET_GAIN) failed");
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return val;
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}
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/* Set filter */
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static void noican_set_filter(int filter)
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{
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if(ioctl(fd_noican, NWZ_NC_SET_FILTER, &filter) < 0)
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panicf("ioctl(NWZ_NC_SET_FILTER) failed");
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}
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/* Get filter */
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static int noican_get_filter(void)
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{
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int val;
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if(ioctl(fd_noican, NWZ_NC_GET_FILTER, &val) < 0)
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panicf("ioctl(NWZ_NC_GET_FILTER) failed");
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return val;
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}
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static void hw_open(void)
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{
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fd_hw = open("/dev/snd/hwC0D0", O_RDWR);
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if(fd_hw < 0)
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panicf("Cannot open '/dev/snd/hwC0D0'");
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}
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static void hw_close(void)
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{
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close(fd_hw);
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}
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/* Acoustic and Cue/Rev control how the volume curve, but it is not clear
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* what the intention of these modes are and the OF does not seem to use
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* them by default */
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bool audiohw_acoustic_enabled(void)
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{
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return alsa_controls_get_bool("CODEC Acoustic Switch");
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}
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void audiohw_enable_acoustic(bool en)
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{
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alsa_controls_set_bool("CODEC Acoustic Switch", en);
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}
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bool audiohw_cuerev_enabled(void)
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{
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return alsa_controls_get_bool("CODEC Cue/Rev Switch");
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}
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void audiohw_enable_cuerev(bool en)
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{
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alsa_controls_set_bool("CODEC Cue/Rev Switch", en);
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}
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void audiohw_preinit(void)
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{
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alsa_controls_init();
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/* turn on codec */
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alsa_controls_set_bool("CODEC Power Switch", true);
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/* mute */
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alsa_controls_set_bool("CODEC Mute Switch", true);
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/* Acoustic and Cue/Rev control how the volume curve, but it is not clear
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* what the intention of these modes are and the OF does not seem to use
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* them by default */
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audiohw_enable_acoustic(false);
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audiohw_enable_cuerev(false);
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/* not sure exactly what it means */
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alsa_controls_set_enum("Playback Src Switch", "Music");
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/* use headphone output */
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alsa_controls_set_enum("Output Switch", "Headphone");
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/* unmute */
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alsa_controls_set_bool("CODEC Mute Switch", false);
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/* sample rate */
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has_sample_rate = alsa_has_control("Sampling Rate");
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/* init noican */
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noican_init();
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if(fd_noican >= 0)
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{
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/* dump configuration, for debug purposes */
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printf("nc hp status: %d\n", noican_get_hp_status());
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printf("nc type: %d\n", noican_get_hp_type());
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printf("nc switch: %d\n", noican_get_switch());
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printf("nc gain: %d\n", noican_get_gain());
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printf("nc filter: %d\n", noican_get_filter());
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/* make sure we start in a clean state */
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noican_set_switch(NWZ_NC_SWITCH_OFF);
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noican_set_hp_type(NC_HP_TYPE_DEFAULT);
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noican_set_filter(NWZ_NC_FILTER_INDEX_0);
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noican_set_gain(NWZ_NC_GAIN_CENTER);
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}
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/* init hw */
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hw_open();
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nwz_codec = find_codec();
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printf("Codec: %s\n", nwz_get_codec_name());
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}
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void audiohw_postinit(void)
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{
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}
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/* volume must be driver unit */
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static void nwz_set_driver_vol(int vol)
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{
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long vols[2];
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/* the driver expects percent, convert from centibel in range 0...x */
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vols[0] = vols[1] = vol;
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/* on some recent players like A10, Sony decided to merge left/right volume
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* into one, thus we need to make sure we write the correct number of values */
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int vol_cnt;
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alsa_controls_get_info("Playback Volume", &vol_cnt);
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alsa_controls_set_ints("Playback Volume", vol_cnt, vols);
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}
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/* volume is in tenth-dB */
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void audiohw_set_volume(int vol_l, int vol_r)
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{
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/* FIXME at the moment we don't support balance and just average left and right.
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* But this could be implemented using pcm alsa digital volume */
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/* the Sony drivers expect vol_l = vol_r */
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int vol = (vol_l + vol_r) / 2;
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printf("request volume %d dB\n", vol / 10);
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struct nwz_vol_curve_t *curve = nwz_get_codec_vol_curve();
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/* min/max for pcm volume */
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int min_pcm = -430;
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int max_pcm = 0;
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/* On some codecs (like cs47L01), Sony clear overdrives the DAC which produces
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* massive clipping at any level (since they fix the DAC volume at around +6dB
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* and then adjust HP volume in negative at the top of range !!). The only
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* solution around this problem is to use the digital volume first so that
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* very quickly the digital volume compensate for the DAC overdrive and we
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* avoid clipping. */
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int sony_clip_level = -80; /* any volume above this will cause massive clipping the DAC */
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/* to avoid the clipping problem, virtually decrease requested volume by the
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* clipping threshold, so that we will compensate in digital later by
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* at least this amount if possibly */
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vol -= sony_clip_level;
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int drv_vol = curve->count - 1;
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/* pick driver level just above request volume */
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while(drv_vol > 0 && curve->level[drv_vol - 1] >= vol)
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drv_vol--;
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/* now remove the artifical volume change */
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vol += sony_clip_level;
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/* now adjust digital volume */
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vol -= curve->level[drv_vol];
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if(vol < min_pcm)
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{
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vol = min_pcm; /* digital cannot do <43dB */
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drv_vol = 0; /* mute */
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}
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else if(vol > max_pcm)
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vol = max_pcm; /* digital cannot do >0dB */
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printf(" set driver volume %d (%d dB)\n", drv_vol, curve->level[drv_vol] / 10);
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nwz_set_driver_vol(drv_vol);
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printf(" set digital volume %d dB\n", vol / 10);
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pcm_alsa_set_digital_volume(vol / 10);
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}
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void audiohw_close(void)
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{
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hw_close();
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alsa_controls_close();
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noican_close();
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}
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void audiohw_set_frequency(int fsel)
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{
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if(has_sample_rate)
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{
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/* it's slightly annoying that Sony put the value in an enum with strings... */
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char freq_str[16];
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sprintf(freq_str, "%d", fsel);
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alsa_controls_set_enum("Sampling Rate", freq_str);
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}
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}
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