mpd/src/output/shout_output_plugin.c

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/*
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* 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.
*/
#include "config.h"
#include "shout_output_plugin.h"
#include "output_api.h"
#include "encoder_plugin.h"
#include "encoder_list.h"
#include "mpd_error.h"
#include <shout/shout.h>
#include <glib.h>
#include <assert.h>
#include <stdlib.h>
#include <stdio.h>
#undef G_LOG_DOMAIN
#define G_LOG_DOMAIN "shout"
#define DEFAULT_CONN_TIMEOUT 2
struct shout_data {
struct audio_output base;
shout_t *shout_conn;
shout_metadata_t *shout_meta;
struct encoder *encoder;
float quality;
int bitrate;
int timeout;
uint8_t buffer[32768];
};
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static int shout_init_count;
shout: introduce pluggable encoder API I've perhaps gone a bit overboard, but here's the current rundown: Both Ogg and MP3 use the "shout" audio output plugin. The shout audio output plugin itself has two new plugins, one for the Ogg encoder, and another for the MP3 (LAME) encoder. Configuration for an Ogg stream doesn't change. For an MP3 stream, configuration is the same as Ogg, with two exceptions. First, you must specify the optional "encoding" parameter, which should be set to "mp3". See mpd.conf(5) for more details. Second, the "quality" parameter is reversed for LAME, such that 1 is high quality for LAME, whereas 10 is high quality for Ogg. I've decomposed the code so that all libshout related operations are done in audioOutput_shout.c, all Ogg specific functions are in audioOutput_shout_ogg.c, and of course then all LAME specific functions are handled in audioOutput_shout_mp3.c. To develop encoder plugins for the shout audio output plugin, I basically just mimicked the plugin system used for audio outputs. This might be overkill, but hopefully if anyone ever wants to support some other sort of stream, like maybe AAC, FLAC, or WMA (hey it could happen), they will hopefully be all set. The Ogg encoder is slightly less optimal under this configuration. It used to send shout data directly out of its ogg_page structures. Now, in the interest of encapsulation, it copies the data from its ogg_page structures into a buffer provided by the shout audio output plugin (see audioOutput_shout_ogg.c, line 77.) I suspect the performance impact is negligible. As for metadata, I'm pretty sure they'll both work. I wrote up a test scaffold that would create a fake tag, and tell the plugin to send it out to the stream every few seconds. It seemed to work fine. Of course, if something does break, I'll be glad to fix it. Lastly, I've renamed lots of things into snake_case, in keeping with normalperson's wishes in that regard. [mk: moved the MP3 patch after this one. Splitted this patch into several parts; the others were already applied before this one. Fixed a bunch GCC warnings and wrong whitespace modifications. Made it compile with mpd-mk by adapting to its prototypes]
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/**
* The quark used for GError.domain.
*/
static inline GQuark
shout_output_quark(void)
{
return g_quark_from_static_string("shout_output");
}
static const struct encoder_plugin *
shout_encoder_plugin_get(const char *name)
shout: introduce pluggable encoder API I've perhaps gone a bit overboard, but here's the current rundown: Both Ogg and MP3 use the "shout" audio output plugin. The shout audio output plugin itself has two new plugins, one for the Ogg encoder, and another for the MP3 (LAME) encoder. Configuration for an Ogg stream doesn't change. For an MP3 stream, configuration is the same as Ogg, with two exceptions. First, you must specify the optional "encoding" parameter, which should be set to "mp3". See mpd.conf(5) for more details. Second, the "quality" parameter is reversed for LAME, such that 1 is high quality for LAME, whereas 10 is high quality for Ogg. I've decomposed the code so that all libshout related operations are done in audioOutput_shout.c, all Ogg specific functions are in audioOutput_shout_ogg.c, and of course then all LAME specific functions are handled in audioOutput_shout_mp3.c. To develop encoder plugins for the shout audio output plugin, I basically just mimicked the plugin system used for audio outputs. This might be overkill, but hopefully if anyone ever wants to support some other sort of stream, like maybe AAC, FLAC, or WMA (hey it could happen), they will hopefully be all set. The Ogg encoder is slightly less optimal under this configuration. It used to send shout data directly out of its ogg_page structures. Now, in the interest of encapsulation, it copies the data from its ogg_page structures into a buffer provided by the shout audio output plugin (see audioOutput_shout_ogg.c, line 77.) I suspect the performance impact is negligible. As for metadata, I'm pretty sure they'll both work. I wrote up a test scaffold that would create a fake tag, and tell the plugin to send it out to the stream every few seconds. It seemed to work fine. Of course, if something does break, I'll be glad to fix it. Lastly, I've renamed lots of things into snake_case, in keeping with normalperson's wishes in that regard. [mk: moved the MP3 patch after this one. Splitted this patch into several parts; the others were already applied before this one. Fixed a bunch GCC warnings and wrong whitespace modifications. Made it compile with mpd-mk by adapting to its prototypes]
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{
if (strcmp(name, "ogg") == 0)
name = "vorbis";
else if (strcmp(name, "mp3") == 0)
name = "lame";
shout: introduce pluggable encoder API I've perhaps gone a bit overboard, but here's the current rundown: Both Ogg and MP3 use the "shout" audio output plugin. The shout audio output plugin itself has two new plugins, one for the Ogg encoder, and another for the MP3 (LAME) encoder. Configuration for an Ogg stream doesn't change. For an MP3 stream, configuration is the same as Ogg, with two exceptions. First, you must specify the optional "encoding" parameter, which should be set to "mp3". See mpd.conf(5) for more details. Second, the "quality" parameter is reversed for LAME, such that 1 is high quality for LAME, whereas 10 is high quality for Ogg. I've decomposed the code so that all libshout related operations are done in audioOutput_shout.c, all Ogg specific functions are in audioOutput_shout_ogg.c, and of course then all LAME specific functions are handled in audioOutput_shout_mp3.c. To develop encoder plugins for the shout audio output plugin, I basically just mimicked the plugin system used for audio outputs. This might be overkill, but hopefully if anyone ever wants to support some other sort of stream, like maybe AAC, FLAC, or WMA (hey it could happen), they will hopefully be all set. The Ogg encoder is slightly less optimal under this configuration. It used to send shout data directly out of its ogg_page structures. Now, in the interest of encapsulation, it copies the data from its ogg_page structures into a buffer provided by the shout audio output plugin (see audioOutput_shout_ogg.c, line 77.) I suspect the performance impact is negligible. As for metadata, I'm pretty sure they'll both work. I wrote up a test scaffold that would create a fake tag, and tell the plugin to send it out to the stream every few seconds. It seemed to work fine. Of course, if something does break, I'll be glad to fix it. Lastly, I've renamed lots of things into snake_case, in keeping with normalperson's wishes in that regard. [mk: moved the MP3 patch after this one. Splitted this patch into several parts; the others were already applied before this one. Fixed a bunch GCC warnings and wrong whitespace modifications. Made it compile with mpd-mk by adapting to its prototypes]
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return encoder_plugin_get(name);
shout: introduce pluggable encoder API I've perhaps gone a bit overboard, but here's the current rundown: Both Ogg and MP3 use the "shout" audio output plugin. The shout audio output plugin itself has two new plugins, one for the Ogg encoder, and another for the MP3 (LAME) encoder. Configuration for an Ogg stream doesn't change. For an MP3 stream, configuration is the same as Ogg, with two exceptions. First, you must specify the optional "encoding" parameter, which should be set to "mp3". See mpd.conf(5) for more details. Second, the "quality" parameter is reversed for LAME, such that 1 is high quality for LAME, whereas 10 is high quality for Ogg. I've decomposed the code so that all libshout related operations are done in audioOutput_shout.c, all Ogg specific functions are in audioOutput_shout_ogg.c, and of course then all LAME specific functions are handled in audioOutput_shout_mp3.c. To develop encoder plugins for the shout audio output plugin, I basically just mimicked the plugin system used for audio outputs. This might be overkill, but hopefully if anyone ever wants to support some other sort of stream, like maybe AAC, FLAC, or WMA (hey it could happen), they will hopefully be all set. The Ogg encoder is slightly less optimal under this configuration. It used to send shout data directly out of its ogg_page structures. Now, in the interest of encapsulation, it copies the data from its ogg_page structures into a buffer provided by the shout audio output plugin (see audioOutput_shout_ogg.c, line 77.) I suspect the performance impact is negligible. As for metadata, I'm pretty sure they'll both work. I wrote up a test scaffold that would create a fake tag, and tell the plugin to send it out to the stream every few seconds. It seemed to work fine. Of course, if something does break, I'll be glad to fix it. Lastly, I've renamed lots of things into snake_case, in keeping with normalperson's wishes in that regard. [mk: moved the MP3 patch after this one. Splitted this patch into several parts; the others were already applied before this one. Fixed a bunch GCC warnings and wrong whitespace modifications. Made it compile with mpd-mk by adapting to its prototypes]
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}
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static struct shout_data *new_shout_data(void)
{
struct shout_data *ret = g_new(struct shout_data, 1);
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ret->shout_conn = shout_new();
ret->shout_meta = shout_metadata_new();
ret->bitrate = -1;
ret->quality = -2.0;
ret->timeout = DEFAULT_CONN_TIMEOUT;
return ret;
}
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static void free_shout_data(struct shout_data *sd)
{
if (sd->shout_meta)
shout_metadata_free(sd->shout_meta);
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if (sd->shout_conn)
shout_free(sd->shout_conn);
g_free(sd);
}
gcc_pure
static const char *
require_block_string(const struct config_param *param, const char *name)
{
const char *value = config_get_block_string(param, name, NULL);
if (value == NULL)
MPD_ERROR("no \"%s\" defined for shout device defined at line " \
"%i\n", name, param->line); \
return value;
}
static bool
my_shout_configure(struct shout_data *sd, const struct config_param *param,
GError **error)
{
const struct audio_format *audio_format =
&sd->base.config_audio_format;
if (!audio_format_fully_defined(audio_format)) {
g_set_error(error, shout_output_quark(), 0,
"Need full audio format specification");
ao_base_finish(&sd->base);
free_shout_data(sd);
return NULL;
}
const char *host = require_block_string(param, "host");
const char *mount = require_block_string(param, "mount");
unsigned port = config_get_block_unsigned(param, "port", 0);
if (port == 0) {
g_set_error(error, shout_output_quark(), 0,
"shout port must be configured");
return false;
}
const char *passwd = require_block_string(param, "password");
const char *name = require_block_string(param, "name");
bool public = config_get_block_bool(param, "public", false);
const char *user = config_get_block_string(param, "user", "source");
const char *value = config_get_block_string(param, "quality", NULL);
if (value != NULL) {
char *test;
sd->quality = strtod(value, &test);
if (*test != '\0' || sd->quality < -1.0 || sd->quality > 10.0) {
g_set_error(error, shout_output_quark(), 0,
"shout quality \"%s\" is not a number in the "
"range -1 to 10, line %i",
value, param->line);
return false;
}
if (config_get_block_string(param, "bitrate", NULL) != NULL) {
g_set_error(error, shout_output_quark(), 0,
"quality and bitrate are "
"both defined");
return false;
}
} else {
value = config_get_block_string(param, "bitrate", NULL);
if (value == NULL) {
g_set_error(error, shout_output_quark(), 0,
"neither bitrate nor quality defined");
return false;
}
char *test;
sd->bitrate = strtol(value, &test, 10);
if (*test != '\0' || sd->bitrate <= 0) {
g_set_error(error, shout_output_quark(), 0,
"bitrate must be a positive integer");
return false;
}
}
const char *encoding = config_get_block_string(param, "encoding",
"ogg");
const struct encoder_plugin *encoder_plugin =
shout_encoder_plugin_get(encoding);
if (encoder_plugin == NULL) {
g_set_error(error, shout_output_quark(), 0,
"couldn't find shout encoder plugin \"%s\"",
encoding);
return false;
}
shout: introduce pluggable encoder API I've perhaps gone a bit overboard, but here's the current rundown: Both Ogg and MP3 use the "shout" audio output plugin. The shout audio output plugin itself has two new plugins, one for the Ogg encoder, and another for the MP3 (LAME) encoder. Configuration for an Ogg stream doesn't change. For an MP3 stream, configuration is the same as Ogg, with two exceptions. First, you must specify the optional "encoding" parameter, which should be set to "mp3". See mpd.conf(5) for more details. Second, the "quality" parameter is reversed for LAME, such that 1 is high quality for LAME, whereas 10 is high quality for Ogg. I've decomposed the code so that all libshout related operations are done in audioOutput_shout.c, all Ogg specific functions are in audioOutput_shout_ogg.c, and of course then all LAME specific functions are handled in audioOutput_shout_mp3.c. To develop encoder plugins for the shout audio output plugin, I basically just mimicked the plugin system used for audio outputs. This might be overkill, but hopefully if anyone ever wants to support some other sort of stream, like maybe AAC, FLAC, or WMA (hey it could happen), they will hopefully be all set. The Ogg encoder is slightly less optimal under this configuration. It used to send shout data directly out of its ogg_page structures. Now, in the interest of encapsulation, it copies the data from its ogg_page structures into a buffer provided by the shout audio output plugin (see audioOutput_shout_ogg.c, line 77.) I suspect the performance impact is negligible. As for metadata, I'm pretty sure they'll both work. I wrote up a test scaffold that would create a fake tag, and tell the plugin to send it out to the stream every few seconds. It seemed to work fine. Of course, if something does break, I'll be glad to fix it. Lastly, I've renamed lots of things into snake_case, in keeping with normalperson's wishes in that regard. [mk: moved the MP3 patch after this one. Splitted this patch into several parts; the others were already applied before this one. Fixed a bunch GCC warnings and wrong whitespace modifications. Made it compile with mpd-mk by adapting to its prototypes]
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sd->encoder = encoder_init(encoder_plugin, param, error);
if (sd->encoder == NULL)
return false;
unsigned shout_format;
if (strcmp(encoding, "mp3") == 0 || strcmp(encoding, "lame") == 0)
shout_format = SHOUT_FORMAT_MP3;
else
shout_format = SHOUT_FORMAT_OGG;
unsigned protocol;
value = config_get_block_string(param, "protocol", NULL);
if (value != NULL) {
if (0 == strcmp(value, "shoutcast") &&
0 != strcmp(encoding, "mp3")) {
g_set_error(error, shout_output_quark(), 0,
"you cannot stream \"%s\" to shoutcast, use mp3",
encoding);
return false;
} else if (0 == strcmp(value, "shoutcast"))
protocol = SHOUT_PROTOCOL_ICY;
else if (0 == strcmp(value, "icecast1"))
protocol = SHOUT_PROTOCOL_XAUDIOCAST;
else if (0 == strcmp(value, "icecast2"))
protocol = SHOUT_PROTOCOL_HTTP;
else {
g_set_error(error, shout_output_quark(), 0,
"shout protocol \"%s\" is not \"shoutcast\" or "
"\"icecast1\"or \"icecast2\"",
value);
return false;
}
} else {
protocol = SHOUT_PROTOCOL_HTTP;
}
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if (shout_set_host(sd->shout_conn, host) != SHOUTERR_SUCCESS ||
shout_set_port(sd->shout_conn, port) != SHOUTERR_SUCCESS ||
shout_set_password(sd->shout_conn, passwd) != SHOUTERR_SUCCESS ||
shout_set_mount(sd->shout_conn, mount) != SHOUTERR_SUCCESS ||
shout_set_name(sd->shout_conn, name) != SHOUTERR_SUCCESS ||
shout_set_user(sd->shout_conn, user) != SHOUTERR_SUCCESS ||
shout_set_public(sd->shout_conn, public) != SHOUTERR_SUCCESS ||
shout_set_format(sd->shout_conn, shout_format)
!= SHOUTERR_SUCCESS ||
shout_set_protocol(sd->shout_conn, protocol) != SHOUTERR_SUCCESS ||
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shout_set_agent(sd->shout_conn, "MPD") != SHOUTERR_SUCCESS) {
g_set_error(error, shout_output_quark(), 0,
"%s", shout_get_error(sd->shout_conn));
return false;
}
/* optional paramters */
sd->timeout = config_get_block_unsigned(param, "timeout",
DEFAULT_CONN_TIMEOUT);
value = config_get_block_string(param, "genre", NULL);
if (value != NULL && shout_set_genre(sd->shout_conn, value)) {
g_set_error(error, shout_output_quark(), 0,
"%s", shout_get_error(sd->shout_conn));
return false;
}
value = config_get_block_string(param, "description", NULL);
if (value != NULL && shout_set_description(sd->shout_conn, value)) {
g_set_error(error, shout_output_quark(), 0,
"%s", shout_get_error(sd->shout_conn));
return false;
}
value = config_get_block_string(param, "url", NULL);
if (value != NULL && shout_set_url(sd->shout_conn, value)) {
g_set_error(error, shout_output_quark(), 0,
"%s", shout_get_error(sd->shout_conn));
return false;
}
{
char temp[11];
memset(temp, 0, sizeof(temp));
snprintf(temp, sizeof(temp), "%u", audio_format->channels);
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shout_set_audio_info(sd->shout_conn, SHOUT_AI_CHANNELS, temp);
snprintf(temp, sizeof(temp), "%u", audio_format->sample_rate);
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shout_set_audio_info(sd->shout_conn, SHOUT_AI_SAMPLERATE, temp);
if (sd->quality >= -1.0) {
snprintf(temp, sizeof(temp), "%2.2f", sd->quality);
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shout_set_audio_info(sd->shout_conn, SHOUT_AI_QUALITY,
temp);
} else {
snprintf(temp, sizeof(temp), "%d", sd->bitrate);
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shout_set_audio_info(sd->shout_conn, SHOUT_AI_BITRATE,
temp);
}
}
return true;
}
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static struct audio_output *
my_shout_init_driver(const struct config_param *param,
GError **error)
{
struct shout_data *sd = new_shout_data();
if (!ao_base_init(&sd->base, &shout_output_plugin, param, error)) {
free_shout_data(sd);
return NULL;
}
if (!my_shout_configure(sd, param, error)) {
ao_base_finish(&sd->base);
free_shout_data(sd);
return NULL;
}
if (shout_init_count == 0)
shout_init();
shout_init_count++;
return &sd->base;
}
static bool
handle_shout_error(struct shout_data *sd, int err, GError **error)
{
switch (err) {
case SHOUTERR_SUCCESS:
break;
case SHOUTERR_UNCONNECTED:
case SHOUTERR_SOCKET:
g_set_error(error, shout_output_quark(), err,
"Lost shout connection to %s:%i: %s",
shout_get_host(sd->shout_conn),
shout_get_port(sd->shout_conn),
shout_get_error(sd->shout_conn));
return false;
default:
g_set_error(error, shout_output_quark(), err,
"connection to %s:%i error: %s",
shout_get_host(sd->shout_conn),
shout_get_port(sd->shout_conn),
shout_get_error(sd->shout_conn));
return false;
}
return true;
}
static bool
write_page(struct shout_data *sd, GError **error)
{
assert(sd->encoder != NULL);
while (true) {
size_t nbytes = encoder_read(sd->encoder,
sd->buffer, sizeof(sd->buffer));
if (nbytes == 0)
return true;
int err = shout_send(sd->shout_conn, sd->buffer, nbytes);
if (!handle_shout_error(sd, err, error))
return false;
}
return true;
}
shout: introduce pluggable encoder API I've perhaps gone a bit overboard, but here's the current rundown: Both Ogg and MP3 use the "shout" audio output plugin. The shout audio output plugin itself has two new plugins, one for the Ogg encoder, and another for the MP3 (LAME) encoder. Configuration for an Ogg stream doesn't change. For an MP3 stream, configuration is the same as Ogg, with two exceptions. First, you must specify the optional "encoding" parameter, which should be set to "mp3". See mpd.conf(5) for more details. Second, the "quality" parameter is reversed for LAME, such that 1 is high quality for LAME, whereas 10 is high quality for Ogg. I've decomposed the code so that all libshout related operations are done in audioOutput_shout.c, all Ogg specific functions are in audioOutput_shout_ogg.c, and of course then all LAME specific functions are handled in audioOutput_shout_mp3.c. To develop encoder plugins for the shout audio output plugin, I basically just mimicked the plugin system used for audio outputs. This might be overkill, but hopefully if anyone ever wants to support some other sort of stream, like maybe AAC, FLAC, or WMA (hey it could happen), they will hopefully be all set. The Ogg encoder is slightly less optimal under this configuration. It used to send shout data directly out of its ogg_page structures. Now, in the interest of encapsulation, it copies the data from its ogg_page structures into a buffer provided by the shout audio output plugin (see audioOutput_shout_ogg.c, line 77.) I suspect the performance impact is negligible. As for metadata, I'm pretty sure they'll both work. I wrote up a test scaffold that would create a fake tag, and tell the plugin to send it out to the stream every few seconds. It seemed to work fine. Of course, if something does break, I'll be glad to fix it. Lastly, I've renamed lots of things into snake_case, in keeping with normalperson's wishes in that regard. [mk: moved the MP3 patch after this one. Splitted this patch into several parts; the others were already applied before this one. Fixed a bunch GCC warnings and wrong whitespace modifications. Made it compile with mpd-mk by adapting to its prototypes]
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static void close_shout_conn(struct shout_data * sd)
{
if (sd->encoder != NULL) {
if (encoder_end(sd->encoder, NULL))
write_page(sd, NULL);
encoder_close(sd->encoder);
}
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if (shout_get_connected(sd->shout_conn) != SHOUTERR_UNCONNECTED &&
shout_close(sd->shout_conn) != SHOUTERR_SUCCESS) {
g_warning("problem closing connection to shout server: %s\n",
shout_get_error(sd->shout_conn));
}
}
static void
my_shout_finish_driver(struct audio_output *ao)
{
struct shout_data *sd = (struct shout_data *)ao;
encoder_finish(sd->encoder);
ao_base_finish(&sd->base);
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free_shout_data(sd);
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shout_init_count--;
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if (shout_init_count == 0)
shout_shutdown();
}
static void
my_shout_drop_buffered_audio(struct audio_output *ao)
{
G_GNUC_UNUSED
struct shout_data *sd = (struct shout_data *)ao;
/* needs to be implemented for shout */
}
static void
my_shout_close_device(struct audio_output *ao)
{
struct shout_data *sd = (struct shout_data *)ao;
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close_shout_conn(sd);
}
static bool
shout_connect(struct shout_data *sd, GError **error)
{
switch (shout_open(sd->shout_conn)) {
case SHOUTERR_SUCCESS:
case SHOUTERR_CONNECTED:
return true;
default:
g_set_error(error, shout_output_quark(), 0,
"problem opening connection to shout server %s:%i: %s",
shout_get_host(sd->shout_conn),
shout_get_port(sd->shout_conn),
shout_get_error(sd->shout_conn));
return false;
}
}
static bool
my_shout_open_device(struct audio_output *ao, struct audio_format *audio_format,
GError **error)
{
struct shout_data *sd = (struct shout_data *)ao;
if (!shout_connect(sd, error))
return false;
if (!encoder_open(sd->encoder, audio_format, error)) {
shout_close(sd->shout_conn);
return false;
}
if (!write_page(sd, error)) {
encoder_close(sd->encoder);
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shout_close(sd->shout_conn);
return false;
}
return true;
}
static unsigned
my_shout_delay(struct audio_output *ao)
{
struct shout_data *sd = (struct shout_data *)ao;
int delay = shout_delay(sd->shout_conn);
if (delay < 0)
delay = 0;
return delay;
}
static size_t
my_shout_play(struct audio_output *ao, const void *chunk, size_t size,
GError **error)
{
struct shout_data *sd = (struct shout_data *)ao;
return encoder_write(sd->encoder, chunk, size, error) &&
write_page(sd, error)
? size
: 0;
}
static bool
my_shout_pause(struct audio_output *ao)
{
static const char silence[1020];
return my_shout_play(ao, silence, sizeof(silence), NULL);
}
static void
shout_tag_to_metadata(const struct tag *tag, char *dest, size_t size)
{
char artist[size];
char title[size];
artist[0] = 0;
title[0] = 0;
for (unsigned i = 0; i < tag->num_items; i++) {
switch (tag->items[i]->type) {
case TAG_ARTIST:
strncpy(artist, tag->items[i]->value, size);
break;
case TAG_TITLE:
strncpy(title, tag->items[i]->value, size);
break;
default:
break;
}
}
snprintf(dest, size, "%s - %s", artist, title);
}
static void my_shout_set_tag(struct audio_output *ao,
2008-09-12 14:01:45 +02:00
const struct tag *tag)
{
struct shout_data *sd = (struct shout_data *)ao;
GError *error = NULL;
if (sd->encoder->plugin->tag != NULL) {
/* encoder plugin supports stream tags */
if (!encoder_pre_tag(sd->encoder, &error)) {
g_warning("%s", error->message);
g_error_free(error);
return;
}
if (!write_page(sd, NULL))
return;
if (!encoder_tag(sd->encoder, tag, &error)) {
g_warning("%s", error->message);
g_error_free(error);
}
} else {
/* no stream tag support: fall back to icy-metadata */
char song[1024];
shout_tag_to_metadata(tag, song, sizeof(song));
shout_metadata_add(sd->shout_meta, "song", song);
if (SHOUTERR_SUCCESS != shout_set_metadata(sd->shout_conn,
sd->shout_meta)) {
g_warning("error setting shout metadata\n");
}
}
write_page(sd, NULL);
}
const struct audio_output_plugin shout_output_plugin = {
.name = "shout",
.init = my_shout_init_driver,
.finish = my_shout_finish_driver,
.open = my_shout_open_device,
.delay = my_shout_delay,
.play = my_shout_play,
.pause = my_shout_pause,
.cancel = my_shout_drop_buffered_audio,
.close = my_shout_close_device,
.send_tag = my_shout_set_tag,
};