mpd/src/output/plugins/ShoutOutputPlugin.cxx

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/*
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* Copyright 2003-2017 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"
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#include "ShoutOutputPlugin.hxx"
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#include "../OutputAPI.hxx"
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#include "../Wrapper.hxx"
#include "encoder/EncoderInterface.hxx"
#include "encoder/EncoderPlugin.hxx"
#include "encoder/EncoderList.hxx"
#include "util/RuntimeError.hxx"
#include "util/Domain.hxx"
#include "Log.hxx"
#include <shout/shout.h>
#include <stdexcept>
#include <assert.h>
#include <stdlib.h>
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#include <string.h>
#include <stdio.h>
static constexpr unsigned DEFAULT_CONN_TIMEOUT = 2;
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struct ShoutOutput final {
FilteredAudioOutput base;
shout_t *shout_conn;
shout_metadata_t *shout_meta;
PreparedEncoder *prepared_encoder = nullptr;
Encoder *encoder;
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float quality = -2.0;
int bitrate = -1;
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int timeout = DEFAULT_CONN_TIMEOUT;
uint8_t buffer[32768];
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explicit ShoutOutput(const ConfigBlock &block);
~ShoutOutput();
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static ShoutOutput *Create(EventLoop &event_loop,
const ConfigBlock &block);
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void Open(AudioFormat &audio_format);
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void Close();
std::chrono::steady_clock::duration Delay() const noexcept;
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void SendTag(const Tag &tag);
size_t Play(const void *chunk, size_t size);
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void Cancel();
bool Pause();
};
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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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static constexpr Domain shout_output_domain("shout_output");
static const char *
require_block_string(const ConfigBlock &block, const char *name)
{
const char *value = block.GetBlockValue(name);
if (value == nullptr)
throw FormatRuntimeError("no \"%s\" defined for shout device defined "
"at line %d\n", name, block.line);
return value;
}
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static const EncoderPlugin *
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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}
ShoutOutput::ShoutOutput(const ConfigBlock &block)
:base(shout_output_plugin, block),
shout_conn(shout_new()),
shout_meta(shout_metadata_new())
{
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const AudioFormat audio_format = base.config_audio_format;
if (!audio_format.IsFullyDefined())
throw std::runtime_error("Need full audio format specification");
const char *host = require_block_string(block, "host");
const char *mount = require_block_string(block, "mount");
unsigned port = block.GetBlockValue("port", 0u);
if (port == 0)
throw std::runtime_error("shout port must be configured");
const char *passwd = require_block_string(block, "password");
const char *name = require_block_string(block, "name");
bool is_public = block.GetBlockValue("public", false);
const char *user = block.GetBlockValue("user", "source");
const char *value = block.GetBlockValue("quality");
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if (value != nullptr) {
char *test;
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quality = strtod(value, &test);
if (*test != '\0' || quality < -1.0 || quality > 10.0)
throw FormatRuntimeError("shout quality \"%s\" is not a number in the "
"range -1 to 10",
value);
if (block.GetBlockValue("bitrate") != nullptr)
throw std::runtime_error("quality and bitrate are "
"both defined");
} else {
value = block.GetBlockValue("bitrate");
if (value == nullptr)
throw std::runtime_error("neither bitrate nor quality defined");
char *test;
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bitrate = strtol(value, &test, 10);
if (*test != '\0' || bitrate <= 0)
throw std::runtime_error("bitrate must be a positive integer");
}
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const char *encoding = block.GetBlockValue("encoder", nullptr);
if (encoding == nullptr)
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encoding = block.GetBlockValue("encoding", "vorbis");
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const auto encoder_plugin = shout_encoder_plugin_get(encoding);
if (encoder_plugin == nullptr)
throw FormatRuntimeError("couldn't find shout encoder plugin \"%s\"",
encoding);
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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prepared_encoder = encoder_init(*encoder_plugin, block);
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 = block.GetBlockValue("protocol");
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if (value != nullptr) {
if (0 == strcmp(value, "shoutcast") &&
0 != strcmp(encoding, "mp3"))
throw FormatRuntimeError("you cannot stream \"%s\" to shoutcast, use mp3",
encoding);
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
throw FormatRuntimeError("shout protocol \"%s\" is not \"shoutcast\" or "
"\"icecast1\"or \"icecast2\"",
value);
} else {
protocol = SHOUT_PROTOCOL_HTTP;
}
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if (shout_set_host(shout_conn, host) != SHOUTERR_SUCCESS ||
shout_set_port(shout_conn, port) != SHOUTERR_SUCCESS ||
shout_set_password(shout_conn, passwd) != SHOUTERR_SUCCESS ||
shout_set_mount(shout_conn, mount) != SHOUTERR_SUCCESS ||
shout_set_name(shout_conn, name) != SHOUTERR_SUCCESS ||
shout_set_user(shout_conn, user) != SHOUTERR_SUCCESS ||
shout_set_public(shout_conn, is_public) != SHOUTERR_SUCCESS ||
shout_set_format(shout_conn, shout_format)
!= SHOUTERR_SUCCESS ||
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shout_set_protocol(shout_conn, protocol) != SHOUTERR_SUCCESS ||
shout_set_agent(shout_conn, "MPD") != SHOUTERR_SUCCESS)
throw std::runtime_error(shout_get_error(shout_conn));
/* optional paramters */
timeout = block.GetBlockValue("timeout", DEFAULT_CONN_TIMEOUT);
value = block.GetBlockValue("genre");
if (value != nullptr && shout_set_genre(shout_conn, value))
throw std::runtime_error(shout_get_error(shout_conn));
value = block.GetBlockValue("description");
if (value != nullptr && shout_set_description(shout_conn, value))
throw std::runtime_error(shout_get_error(shout_conn));
value = block.GetBlockValue("url");
if (value != nullptr && shout_set_url(shout_conn, value))
throw std::runtime_error(shout_get_error(shout_conn));
{
char temp[11];
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snprintf(temp, sizeof(temp), "%u", audio_format.channels);
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shout_set_audio_info(shout_conn, SHOUT_AI_CHANNELS, temp);
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snprintf(temp, sizeof(temp), "%u", audio_format.sample_rate);
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shout_set_audio_info(shout_conn, SHOUT_AI_SAMPLERATE, temp);
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if (quality >= -1.0) {
snprintf(temp, sizeof(temp), "%2.2f", quality);
shout_set_audio_info(shout_conn, SHOUT_AI_QUALITY,
temp);
} else {
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snprintf(temp, sizeof(temp), "%d", bitrate);
shout_set_audio_info(shout_conn, SHOUT_AI_BITRATE,
temp);
}
}
}
ShoutOutput::~ShoutOutput()
{
if (shout_meta != nullptr)
shout_metadata_free(shout_meta);
if (shout_conn != nullptr)
shout_free(shout_conn);
shout_init_count--;
if (shout_init_count == 0)
shout_shutdown();
delete prepared_encoder;
}
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ShoutOutput *
ShoutOutput::Create(EventLoop &, const ConfigBlock &block)
{
if (shout_init_count == 0)
shout_init();
shout_init_count++;
return new ShoutOutput(block);
}
static void
handle_shout_error(ShoutOutput *sd, int err)
{
switch (err) {
case SHOUTERR_SUCCESS:
break;
case SHOUTERR_UNCONNECTED:
case SHOUTERR_SOCKET:
throw FormatRuntimeError("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));
default:
throw FormatRuntimeError("connection to %s:%i error: %s",
shout_get_host(sd->shout_conn),
shout_get_port(sd->shout_conn),
shout_get_error(sd->shout_conn));
}
}
static bool
write_page(ShoutOutput *sd)
{
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assert(sd->encoder != nullptr);
while (true) {
size_t nbytes = sd->encoder->Read(sd->buffer,
sizeof(sd->buffer));
if (nbytes == 0)
return true;
int err = shout_send(sd->shout_conn, sd->buffer, nbytes);
handle_shout_error(sd, err);
}
return true;
}
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void
ShoutOutput::Close()
{
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if (encoder != nullptr) {
try {
encoder->End();
write_page(this);
} catch (const std::runtime_error &) {
/* ignore */
}
delete encoder;
}
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if (shout_get_connected(shout_conn) != SHOUTERR_UNCONNECTED &&
shout_close(shout_conn) != SHOUTERR_SUCCESS) {
FormatWarning(shout_output_domain,
"problem closing connection to shout server: %s",
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shout_get_error(shout_conn));
}
}
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void
ShoutOutput::Cancel()
{
/* needs to be implemented for shout */
}
static void
shout_connect(ShoutOutput *sd)
{
switch (shout_open(sd->shout_conn)) {
case SHOUTERR_SUCCESS:
case SHOUTERR_CONNECTED:
break;
default:
throw FormatRuntimeError("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));
}
}
void
ShoutOutput::Open(AudioFormat &audio_format)
{
shout_connect(this);
try {
encoder = prepared_encoder->Open(audio_format);
try {
write_page(this);
} catch (const std::runtime_error &) {
delete encoder;
throw;
}
} catch (const std::runtime_error &) {
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shout_close(shout_conn);
throw;
}
}
std::chrono::steady_clock::duration
ShoutOutput::Delay() const noexcept
{
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int delay = shout_delay(shout_conn);
if (delay < 0)
delay = 0;
return std::chrono::milliseconds(delay);
}
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size_t
ShoutOutput::Play(const void *chunk, size_t size)
{
encoder->Write(chunk, size);
write_page(this);
return size;
}
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bool
ShoutOutput::Pause()
{
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static char silence[1020];
try {
encoder->Write(silence, sizeof(silence));
write_page(this);
} catch (const std::runtime_error &) {
return false;
}
return true;
}
static void
shout_tag_to_metadata(const Tag &tag, char *dest, size_t size)
{
char artist[size];
char title[size];
artist[0] = 0;
title[0] = 0;
for (const auto &item : tag) {
switch (item.type) {
case TAG_ARTIST:
strncpy(artist, item.value, size);
break;
case TAG_TITLE:
strncpy(title, item.value, size);
break;
default:
break;
}
}
snprintf(dest, size, "%s - %s", artist, title);
}
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void
ShoutOutput::SendTag(const Tag &tag)
{
if (encoder->ImplementsTag()) {
/* encoder plugin supports stream tags */
encoder->PreTag();
write_page(this);
encoder->SendTag(tag);
} else {
/* no stream tag support: fall back to icy-metadata */
char song[1024];
shout_tag_to_metadata(tag, song, sizeof(song));
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shout_metadata_add(shout_meta, "song", song);
if (SHOUTERR_SUCCESS != shout_set_metadata(shout_conn,
shout_meta)) {
LogWarning(shout_output_domain,
"error setting shout metadata");
}
}
write_page(this);
}
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typedef AudioOutputWrapper<ShoutOutput> Wrapper;
const struct AudioOutputPlugin shout_output_plugin = {
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"shout",
nullptr,
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&Wrapper::Init,
&Wrapper::Finish,
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nullptr,
nullptr,
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&Wrapper::Open,
&Wrapper::Close,
&Wrapper::Delay,
&Wrapper::SendTag,
&Wrapper::Play,
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nullptr,
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&Wrapper::Cancel,
&Wrapper::Pause,
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nullptr,
};