/*
* Copyright (C) 2003-2009 The Music Player Daemon Project
* http://www.musicpd.org
*
* 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 program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with this program; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
/*
* Media MVP audio output based on code from MVPMC project:
* http://mvpmc.sourceforge.net/
*/
#include "../output_api.h"
#include <glib.h>
#include <sys/types.h>
#include <sys/stat.h>
#include <sys/ioctl.h>
#include <fcntl.h>
#include <errno.h>
#include <unistd.h>
#include <stdlib.h>
#undef G_LOG_DOMAIN
#define G_LOG_DOMAIN "mvp"
typedef struct {
unsigned long dsp_status;
unsigned long stream_decode_type;
unsigned long sample_rate;
unsigned long bit_rate;
unsigned long raw[64 / sizeof(unsigned long)];
} aud_status_t;
#define MVP_SET_AUD_STOP _IOW('a',1,int)
#define MVP_SET_AUD_PLAY _IOW('a',2,int)
#define MVP_SET_AUD_PAUSE _IOW('a',3,int)
#define MVP_SET_AUD_UNPAUSE _IOW('a',4,int)
#define MVP_SET_AUD_SRC _IOW('a',5,int)
#define MVP_SET_AUD_MUTE _IOW('a',6,int)
#define MVP_SET_AUD_BYPASS _IOW('a',8,int)
#define MVP_SET_AUD_CHANNEL _IOW('a',9,int)
#define MVP_GET_AUD_STATUS _IOR('a',10,aud_status_t)
#define MVP_SET_AUD_VOLUME _IOW('a',13,int)
#define MVP_GET_AUD_VOLUME _IOR('a',14,int)
#define MVP_SET_AUD_STREAMTYPE _IOW('a',15,int)
#define MVP_SET_AUD_FORMAT _IOW('a',16,int)
#define MVP_GET_AUD_SYNC _IOR('a',21,pts_sync_data_t*)
#define MVP_SET_AUD_STC _IOW('a',22,long long int *)
#define MVP_SET_AUD_SYNC _IOW('a',23,int)
#define MVP_SET_AUD_END_STREAM _IOW('a',25,int)
#define MVP_SET_AUD_RESET _IOW('a',26,int)
#define MVP_SET_AUD_DAC_CLK _IOW('a',27,int)
#define MVP_GET_AUD_REGS _IOW('a',28,aud_ctl_regs_t*)
struct mvp_data {
struct audio_format audio_format;
int fd;
};
static const unsigned mvp_sample_rates[][3] = {
{9, 8000, 32000},
{10, 11025, 44100},
{11, 12000, 48000},
{1, 16000, 32000},
{2, 22050, 44100},
{3, 24000, 48000},
{5, 32000, 32000},
{0, 44100, 44100},
{7, 48000, 48000},
{13, 64000, 32000},
{14, 88200, 44100},
{15, 96000, 48000}
};
/**
* The quark used for GError.domain.
*/
static inline GQuark
mvp_output_quark(void)
{
return g_quark_from_static_string("mvp_output");
}
/**
* Translate a sample rate to a MVP sample rate.
*
* @param sample_rate the sample rate in Hz
*/
static unsigned
mvp_find_sample_rate(unsigned sample_rate)
{
for (unsigned i = 0; i < G_N_ELEMENTS(mvp_sample_rates); ++i)
if (mvp_sample_rates[i][1] == sample_rate)
return mvp_sample_rates[i][0];
return (unsigned)-1;
}
static bool
mvp_output_test_default_device(void)
{
int fd;
fd = open("/dev/adec_pcm", O_WRONLY);
if (fd >= 0) {
close(fd);
return true;
}
g_warning("Error opening PCM device \"/dev/adec_pcm\": %s\n",
strerror(errno));
return false;
}
static void *
mvp_output_init(G_GNUC_UNUSED const struct audio_format *audio_format,
G_GNUC_UNUSED const struct config_param *param,
G_GNUC_UNUSED GError **error)
{
struct mvp_data *md = g_new(struct mvp_data, 1);
md->fd = -1;
return md;
}
static void
mvp_output_finish(void *data)
{
struct mvp_data *md = data;
g_free(md);
}
static bool
mvp_set_pcm_params(struct mvp_data *md, struct audio_format *audio_format,
GError **error)
{
unsigned mix[5];
switch (audio_format->channels) {
case 1:
mix[0] = 1;
break;
case 2:
mix[0] = 0;
break;
default:
g_debug("unsupported channel count %u - falling back to stereo",
audio_format->channels);
audio_format->channels = 2;
mix[0] = 0;
break;
}
/* 0,1=24bit(24) , 2,3=16bit */
switch (audio_format->bits) {
case 16:
mix[1] = 2;
break;
case 24:
mix[1] = 0;
break;
default:
g_debug("unsupported sample format %u - falling back to stereo",
audio_format->bits);
audio_format->bits = 16;
mix[1] = 2;
break;
}
mix[3] = 0; /* stream type? */
mix[4] = G_BYTE_ORDER == G_LITTLE_ENDIAN;
/*
* if there is an exact match for the frequency, use it.
*/
mix[2] = mvp_find_sample_rate(audio_format->sample_rate);
if (mix[2] == (unsigned)-1) {
g_set_error(error, mvp_output_quark(), 0,
"Can not find suitable output frequency for %u",
audio_format->sample_rate);
return false;
}
if (ioctl(md->fd, MVP_SET_AUD_FORMAT, &mix) < 0) {
g_set_error(error, mvp_output_quark(), errno,
"Can not set audio format");
return false;
}
if (ioctl(md->fd, MVP_SET_AUD_SYNC, 2) != 0) {
g_set_error(error, mvp_output_quark(), errno,
"Can not set audio sync");
return false;
}
if (ioctl(md->fd, MVP_SET_AUD_PLAY, 0) < 0) {
g_set_error(error, mvp_output_quark(), errno,
"Can not set audio play mode");
return false;
}
return true;
}
static bool
mvp_output_open(void *data, struct audio_format *audio_format, GError **error)
{
struct mvp_data *md = data;
long long int stc = 0;
int mix[5] = { 0, 2, 7, 1, 0 };
bool success;
if ((md->fd = open("/dev/adec_pcm", O_RDWR | O_NONBLOCK)) < 0) {
g_set_error(error, mvp_output_quark(), errno,
"Error opening /dev/adec_pcm: %s",
strerror(errno));
return false;
}
if (ioctl(md->fd, MVP_SET_AUD_SRC, 1) < 0) {
g_set_error(error, mvp_output_quark(), errno,
"Error setting audio source: %s",
strerror(errno));
return false;
}
if (ioctl(md->fd, MVP_SET_AUD_STREAMTYPE, 0) < 0) {
g_set_error(error, mvp_output_quark(), errno,
"Error setting audio streamtype: %s",
strerror(errno));
return false;
}
if (ioctl(md->fd, MVP_SET_AUD_FORMAT, &mix) < 0) {
g_set_error(error, mvp_output_quark(), errno,
"Error setting audio format: %s",
strerror(errno));
return false;
}
ioctl(md->fd, MVP_SET_AUD_STC, &stc);
if (ioctl(md->fd, MVP_SET_AUD_BYPASS, 1) < 0) {
g_set_error(error, mvp_output_quark(), errno,
"Error setting audio streamtype: %s",
strerror(errno));
return false;
}
success = mvp_set_pcm_params(md, audio_format, error);
if (!success)
return false;
md->audio_format = *audio_format;
return true;
}
static void mvp_output_close(void *data)
{
struct mvp_data *md = data;
if (md->fd >= 0)
close(md->fd);
md->fd = -1;
}
static void mvp_output_cancel(void *data)
{
struct mvp_data *md = data;
if (md->fd >= 0) {
ioctl(md->fd, MVP_SET_AUD_RESET, 0x11);
close(md->fd);
md->fd = -1;
}
}
static size_t
mvp_output_play(void *data, const void *chunk, size_t size, GError **error)
{
struct mvp_data *md = data;
ssize_t ret;
/* reopen the device since it was closed by dropBufferedAudio */
if (md->fd < 0) {
bool success;
success = mvp_output_open(md, &md->audio_format, error);
if (!success)
return 0;
}
while (true) {
ret = write(md->fd, chunk, size);
if (ret > 0)
return (size_t)ret;
if (ret < 0) {
if (errno == EINTR)
continue;
g_set_error(error, mvp_output_quark(), errno,
"Failed to write: %s", strerror(errno));
return 0;
}
}
}
const struct audio_output_plugin mvp_output_plugin = {
.name = "mvp",
.test_default_device = mvp_output_test_default_device,
.init = mvp_output_init,
.finish = mvp_output_finish,
.open = mvp_output_open,
.close = mvp_output_close,
.play = mvp_output_play,
.cancel = mvp_output_cancel,
};
