output/mvp: remove obsolete plugin

The hardware is obsolete, and the product does not exist anymore on
the Hauppauge web site.  Let's see if anybody complains about the
removal.
This commit is contained in:
Max Kellermann 2013-04-17 00:33:37 +02:00
parent f492c78e2e
commit cc6c452854
8 changed files with 2 additions and 406 deletions

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@ -37,9 +37,6 @@ Linux. You will need libasound.
FIFO FIFO
This is a mostly undocumented, developer plugin to transmit raw data. This is a mostly undocumented, developer plugin to transmit raw data.
MVP - http://en.wikipedia.org/wiki/Hauppauge_MediaMVP
A network media player.
OSS - http://www.opensound.com OSS - http://www.opensound.com
Open Sound System. Open Sound System.

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@ -842,11 +842,6 @@ liboutput_plugins_a_SOURCES += \
src/output/JackOutputPlugin.cxx src/output/JackOutputPlugin.hxx src/output/JackOutputPlugin.cxx src/output/JackOutputPlugin.hxx
endif endif
if HAVE_MVP
liboutput_plugins_a_SOURCES += \
src/output/mvp_output_plugin.c src/output/mvp_output_plugin.h
endif
if HAVE_OSS if HAVE_OSS
liboutput_plugins_a_SOURCES += \ liboutput_plugins_a_SOURCES += \
src/output/OssOutputPlugin.cxx \ src/output/OssOutputPlugin.cxx \

1
NEWS
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@ -12,6 +12,7 @@ ver 0.18 (2012/??/??)
* output: * output:
- new option "tags" may be used to disable sending tags to output - new option "tags" may be used to disable sending tags to output
- alsa: workaround for noise after manual song change - alsa: workaround for noise after manual song change
- mvp: remove obsolete plugin
* improved decoder/output error reporting * improved decoder/output error reporting
* eliminate timer wakeup on idle MPD * eliminate timer wakeup on idle MPD

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@ -325,11 +325,6 @@ AC_ARG_ENABLE(mpg123,
[enable libmpg123 decoder plugin]),, [enable libmpg123 decoder plugin]),,
enable_mpg123=auto) enable_mpg123=auto)
AC_ARG_ENABLE(mvp,
AS_HELP_STRING([--enable-mvp],
[enable support for Hauppauge Media MVP (default: disable)]),,
enable_mvp=no)
AC_ARG_ENABLE(openal, AC_ARG_ENABLE(openal,
AS_HELP_STRING([--enable-openal], AS_HELP_STRING([--enable-openal],
[enable OpenAL support (default: disable)]),, [enable OpenAL support (default: disable)]),,
@ -1358,13 +1353,6 @@ fi
AM_CONDITIONAL(HAVE_AO, test x$enable_ao = xyes) AM_CONDITIONAL(HAVE_AO, test x$enable_ao = xyes)
dnl ----------------------------------- MVP -----------------------------------
if test x$enable_mvp = xyes; then
AC_DEFINE(HAVE_MVP,1,[Define to enable Hauppauge Media MVP support])
fi
AM_CONDITIONAL(HAVE_MVP, test x$enable_mvp = xyes)
dnl ---------------------------------- OpenAL --------------------------------- dnl ---------------------------------- OpenAL ---------------------------------
AC_SUBST(OPENAL_CFLAGS,"") AC_SUBST(OPENAL_CFLAGS,"")
AC_SUBST(OPENAL_LIBS,"") AC_SUBST(OPENAL_LIBS,"")
@ -1504,7 +1492,6 @@ if
test x$enable_fifo = xno && test x$enable_fifo = xno &&
test x$enable_httpd_output = xno && test x$enable_httpd_output = xno &&
test x$enable_jack = xno && test x$enable_jack = xno &&
test x$enable_mvp = xno; then
test x$enable_openal = xno && test x$enable_openal = xno &&
test x$enable_oss = xno && test x$enable_oss = xno &&
test x$enable_osx = xno && test x$enable_osx = xno &&
@ -1513,7 +1500,7 @@ if
test x$enable_recorder_output = xno && test x$enable_recorder_output = xno &&
test x$enable_shout = xno && test x$enable_shout = xno &&
test x$enable_solaris_output = xno && test x$enable_solaris_output = xno &&
test x$enable_winmm_output = xno && test x$enable_winmm_output = xno; then
AC_MSG_ERROR([No Audio Output types configured!]) AC_MSG_ERROR([No Audio Output types configured!])
fi fi
@ -1662,7 +1649,6 @@ results(httpd_output,[HTTP Daemon])
results(jack,[JACK]) results(jack,[JACK])
printf '\n\t' printf '\n\t'
results(ao,[libao]) results(ao,[libao])
results(mvp, [Media MVP])
results(oss,[OSS]) results(oss,[OSS])
results(openal,[OpenAL]) results(openal,[OpenAL])
results(osx, [OS X]) results(osx, [OS X])

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@ -1492,16 +1492,6 @@ systemctl start mpd.socket</programlisting>
</informaltable> </informaltable>
</section> </section>
<section>
<title><varname>mvp</varname></title>
<para>
The <varname>mvp</varname> plugin uses the proprietary
Hauppauge Media MVP interface. We do not know any user of
this plugin, and we do not know if it actually works.
</para>
</section>
<section> <section>
<title><varname>httpd</varname></title> <title><varname>httpd</varname></title>

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@ -26,7 +26,6 @@
#include "output/FifoOutputPlugin.hxx" #include "output/FifoOutputPlugin.hxx"
#include "output/HttpdOutputPlugin.hxx" #include "output/HttpdOutputPlugin.hxx"
#include "output/JackOutputPlugin.hxx" #include "output/JackOutputPlugin.hxx"
#include "output/mvp_output_plugin.h"
#include "output/NullOutputPlugin.hxx" #include "output/NullOutputPlugin.hxx"
#include "output/openal_output_plugin.h" #include "output/openal_output_plugin.h"
#include "output/OssOutputPlugin.hxx" #include "output/OssOutputPlugin.hxx"
@ -74,9 +73,6 @@ const struct audio_output_plugin *const audio_output_plugins[] = {
#ifdef HAVE_PULSE #ifdef HAVE_PULSE
&pulse_output_plugin, &pulse_output_plugin,
#endif #endif
#ifdef HAVE_MVP
&mvp_output_plugin,
#endif
#ifdef HAVE_JACK #ifdef HAVE_JACK
&jack_output_plugin, &jack_output_plugin,
#endif #endif

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@ -1,344 +0,0 @@
/*
* Copyright (C) 2003-2011 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 "config.h"
#include "mvp_output_plugin.h"
#include "output_api.h"
#include "fd_util.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_output base;
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_cloexec("/dev/adec_pcm", O_WRONLY, 0);
if (fd >= 0) {
close(fd);
return true;
}
g_warning("Error opening PCM device \"/dev/adec_pcm\": %s\n",
g_strerror(errno));
return false;
}
static struct audio_output *
mvp_output_init(G_GNUC_UNUSED const struct config_param *param, GError **error)
{
struct mvp_data *md = g_new(struct mvp_data, 1);
if (!ao_base_init(&md->base, &mvp_output_plugin, param, error)) {
g_free(md);
return NULL;
}
md->fd = -1;
return &md->base;
}
static void
mvp_output_finish(struct audio_output *ao)
{
struct mvp_data *md = (struct mvp_data *)ao;
ao_base_finish(&md->base);
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->format) {
case SAMPLE_FORMAT_S16:
mix[1] = 2;
break;
case SAMPLE_FORMAT_S24_P32:
mix[1] = 0;
break;
default:
g_debug("unsupported sample format %s - falling back to 16 bit",
sample_format_to_string(audio_format->format));
audio_format->format = SAMPLE_FORMAT_S16;
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(struct audio_output *ao, struct audio_format *audio_format,
GError **error)
{
struct mvp_data *md = (struct mvp_data *)ao;
long long int stc = 0;
int mix[5] = { 0, 2, 7, 1, 0 };
bool success;
md->fd = open_cloexec("/dev/adec_pcm", O_RDWR | O_NONBLOCK, 0);
if (md->fd < 0) {
g_set_error(error, mvp_output_quark(), errno,
"Error opening /dev/adec_pcm: %s",
g_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",
g_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",
g_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",
g_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",
g_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(struct audio_output *ao)
{
struct mvp_data *md = (struct mvp_data *)ao;
if (md->fd >= 0)
close(md->fd);
md->fd = -1;
}
static void mvp_output_cancel(struct audio_output *ao)
{
struct mvp_data *md = (struct mvp_data *)ao;
if (md->fd >= 0) {
ioctl(md->fd, MVP_SET_AUD_RESET, 0x11);
close(md->fd);
md->fd = -1;
}
}
static size_t
mvp_output_play(struct audio_output *ao, const void *chunk, size_t size,
GError **error)
{
struct mvp_data *md = (struct mvp_data *)ao;
ssize_t ret;
/* reopen the device since it was closed by dropBufferedAudio */
if (md->fd < 0) {
bool success;
success = mvp_output_open(ao, &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", g_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,
};

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@ -1,25 +0,0 @@
/*
* Copyright (C) 2003-2011 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.
*/
#ifndef MPD_MVP_OUTPUT_PLUGIN_H
#define MPD_MVP_OUTPUT_PLUGIN_H
extern const struct audio_output_plugin mvp_output_plugin;
#endif