1 files changed, 570 insertions, 0 deletions
diff --git a/src/output/HttpdOutputPlugin.cxx b/src/output/HttpdOutputPlugin.cxx
new file mode 100644
index 000000000..cb515e657
--- /dev/null
+++ b/src/output/HttpdOutputPlugin.cxx
@@ -0,0 +1,570 @@
+/*
+ * Copyright (C) 2003-2013 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 "HttpdOutputPlugin.hxx"
+#include "HttpdInternal.hxx"
+#include "HttpdClient.hxx"
+#include "output_api.h"
+#include "encoder_plugin.h"
+#include "encoder_list.h"
+#include "resolver.h"
+#include "Page.hxx"
+#include "IcyMetaDataServer.hxx"
+#include "fd_util.h"
+#include "Main.hxx"
+
+#include <assert.h>
+
+#include <sys/types.h>
+#include <unistd.h>
+#include <errno.h>
+
+#ifdef HAVE_LIBWRAP
+#include <sys/socket.h> /* needed for AF_UNIX */
+#include <tcpd.h>
+#endif
+
+#undef G_LOG_DOMAIN
+#define G_LOG_DOMAIN "httpd_output"
+
+/**
+ * The quark used for GError.domain.
+ */
+static inline GQuark
+httpd_output_quark(void)
+{
+	return g_quark_from_static_string("httpd_output");
+}
+
+inline
+HttpdOutput::HttpdOutput(EventLoop &_loop)
+	:ServerSocket(_loop),
+	 encoder(nullptr), unflushed_input(0),
+	 metadata(nullptr)
+{
+}
+
+HttpdOutput::~HttpdOutput()
+{
+	if (metadata != nullptr)
+		metadata->Unref();
+
+	if (encoder != nullptr)
+		encoder_finish(encoder);
+
+}
+
+inline bool
+HttpdOutput::Bind(GError **error_r)
+{
+	open = false;
+
+	const ScopeLock protect(mutex);
+	return ServerSocket::Open(error_r);
+}
+
+inline void
+HttpdOutput::Unbind()
+{
+	assert(!open);
+
+	const ScopeLock protect(mutex);
+	ServerSocket::Close();
+}
+
+inline bool
+HttpdOutput::Configure(const config_param *param, GError **error_r)
+{
+	/* read configuration */
+	name = config_get_block_string(param, "name", "Set name in config");
+	genre = config_get_block_string(param, "genre", "Set genre in config");
+	website = config_get_block_string(param, "website",
+					  "Set website in config");
+
+	guint port = config_get_block_unsigned(param, "port", 8000);
+
+	const char *encoder_name =
+		config_get_block_string(param, "encoder", "vorbis");
+	const struct encoder_plugin *encoder_plugin =
+		encoder_plugin_get(encoder_name);
+	if (encoder_plugin == NULL) {
+		g_set_error(error_r, httpd_output_quark(), 0,
+			    "No such encoder: %s", encoder_name);
+		return false;
+	}
+
+	clients_max = config_get_block_unsigned(param,"max_clients", 0);
+
+	/* set up bind_to_address */
+
+	const char *bind_to_address =
+		config_get_block_string(param, "bind_to_address", NULL);
+	bool success = bind_to_address != NULL &&
+		strcmp(bind_to_address, "any") != 0
+		? AddHost(bind_to_address, port, error_r)
+		: AddPort(port, error_r);
+	if (!success)
+		return false;
+
+	/* initialize encoder */
+
+	encoder = encoder_init(encoder_plugin, param, error_r);
+	if (encoder == nullptr)
+		return false;
+
+	/* determine content type */
+	content_type = encoder_get_mime_type(encoder);
+	if (content_type == nullptr)
+		content_type = "application/octet-stream";
+
+	return true;
+}
+
+static struct audio_output *
+httpd_output_init(const struct config_param *param,
+		  GError **error_r)
+{
+	HttpdOutput *httpd = new HttpdOutput(*main_loop);
+
+	if (!ao_base_init(&httpd->base, &httpd_output_plugin, param,
+			  error_r)) {
+		delete httpd;
+		return nullptr;
+	}
+
+	if (!httpd->Configure(param, error_r)) {
+		ao_base_finish(&httpd->base);
+		delete httpd;
+		return nullptr;
+	}
+
+	return &httpd->base;
+}
+
+#if GCC_CHECK_VERSION(4,6) || defined(__clang__)
+#pragma GCC diagnostic push
+#pragma GCC diagnostic ignored "-Winvalid-offsetof"
+#endif
+
+static inline constexpr HttpdOutput *
+Cast(audio_output *ao)
+{
+	return (HttpdOutput *)((char *)ao - offsetof(HttpdOutput, base));
+}
+
+#if GCC_CHECK_VERSION(4,6) || defined(__clang__)
+#pragma GCC diagnostic pop
+#endif
+
+static void
+httpd_output_finish(struct audio_output *ao)
+{
+	HttpdOutput *httpd = Cast(ao);
+
+	ao_base_finish(&httpd->base);
+	delete httpd;
+}
+
+/**
+ * Creates a new #HttpdClient object and adds it into the
+ * HttpdOutput.clients linked list.
+ */
+inline void
+HttpdOutput::AddClient(int fd)
+{
+	clients.emplace_front(this, fd, GetEventLoop(),
+			      encoder->plugin->tag == NULL);
+	++clients_cnt;
+
+	/* pass metadata to client */
+	if (metadata != nullptr)
+		clients.front().PushMetaData(metadata);
+}
+
+void
+HttpdOutput::OnAccept(int fd, const sockaddr &address,
+		      size_t address_length, gcc_unused int uid)
+{
+	/* the listener socket has become readable - a client has
+	   connected */
+
+#ifdef HAVE_LIBWRAP
+	if (address.sa_family != AF_UNIX) {
+		char *hostaddr = sockaddr_to_string(&address, address_length, NULL);
+		const char *progname = g_get_prgname();
+
+		struct request_info req;
+		request_init(&req, RQ_FILE, fd, RQ_DAEMON, progname, 0);
+
+		fromhost(&req);
+
+		if (!hosts_access(&req)) {
+			/* tcp wrappers says no */
+			g_warning("libwrap refused connection (libwrap=%s) from %s",
+			      progname, hostaddr);
+			g_free(hostaddr);
+			close_socket(fd);
+			return;
+		}
+
+		g_free(hostaddr);
+	}
+#else
+	(void)address;
+	(void)address_length;
+#endif	/* HAVE_WRAP */
+
+	const ScopeLock protect(mutex);
+
+	if (fd >= 0) {
+		/* can we allow additional client */
+		if (open && (clients_max == 0 ||  clients_cnt < clients_max))
+			AddClient(fd);
+		else
+			close_socket(fd);
+	} else if (fd < 0 && errno != EINTR) {
+		g_warning("accept() failed: %s", g_strerror(errno));
+	}
+}
+
+Page *
+HttpdOutput::ReadPage()
+{
+	if (unflushed_input >= 65536) {
+		/* we have fed a lot of input into the encoder, but it
+		   didn't give anything back yet - flush now to avoid
+		   buffer underruns */
+		encoder_flush(encoder, NULL);
+		unflushed_input = 0;
+	}
+
+	size_t size = 0;
+	do {
+		size_t nbytes = encoder_read(encoder,
+					     buffer + size,
+					     sizeof(buffer) - size);
+		if (nbytes == 0)
+			break;
+
+		unflushed_input = 0;
+
+		size += nbytes;
+	} while (size < sizeof(buffer));
+
+	if (size == 0)
+		return NULL;
+
+	return Page::Copy(buffer, size);
+}
+
+static bool
+httpd_output_enable(struct audio_output *ao, GError **error_r)
+{
+	HttpdOutput *httpd = Cast(ao);
+
+	return httpd->Bind(error_r);
+}
+
+static void
+httpd_output_disable(struct audio_output *ao)
+{
+	HttpdOutput *httpd = Cast(ao);
+
+	httpd->Unbind();
+}
+
+inline bool
+HttpdOutput::OpenEncoder(struct audio_format *audio_format, GError **error)
+{
+	if (!encoder_open(encoder, audio_format, error))
+		return false;
+
+	/* we have to remember the encoder header, i.e. the first
+	   bytes of encoder output after opening it, because it has to
+	   be sent to every new client */
+	header = ReadPage();
+
+	unflushed_input = 0;
+
+	return true;
+}
+
+inline bool
+HttpdOutput::Open(struct audio_format *audio_format, GError **error_r)
+{
+	assert(!open);
+	assert(clients.empty());
+
+	/* open the encoder */
+
+	if (!OpenEncoder(audio_format, error_r))
+		return false;
+
+	/* initialize other attributes */
+
+	clients_cnt = 0;
+	timer = timer_new(audio_format);
+
+	open = true;
+
+	return true;
+}
+
+static bool
+httpd_output_open(struct audio_output *ao, struct audio_format *audio_format,
+		  GError **error)
+{
+	HttpdOutput *httpd = Cast(ao);
+
+	assert(httpd->clients.empty());
+
+	const ScopeLock protect(httpd->mutex);
+	return httpd->Open(audio_format, error);
+}
+
+inline void
+HttpdOutput::Close()
+{
+	assert(open);
+
+	open = false;
+
+	timer_free(timer);
+
+	clients.clear();
+
+	if (header != NULL)
+		header->Unref();
+
+	encoder_close(encoder);
+}
+
+static void
+httpd_output_close(struct audio_output *ao)
+{
+	HttpdOutput *httpd = Cast(ao);
+
+	const ScopeLock protect(httpd->mutex);
+	httpd->Close();
+}
+
+void
+HttpdOutput::RemoveClient(HttpdClient &client)
+{
+	assert(clients_cnt > 0);
+
+	for (auto prev = clients.before_begin(), i = std::next(prev);;
+	     prev = i, i = std::next(prev)) {
+		assert(i != clients.end());
+		if (&*i == &client) {
+			clients.erase_after(prev);
+			clients_cnt--;
+			break;
+		}
+	}
+}
+
+void
+HttpdOutput::SendHeader(HttpdClient &client) const
+{
+	if (header != NULL)
+		client.PushPage(header);
+}
+
+static unsigned
+httpd_output_delay(struct audio_output *ao)
+{
+	HttpdOutput *httpd = Cast(ao);
+
+	if (!httpd->LockHasClients() && httpd->base.pause) {
+		/* if there's no client and this output is paused,
+		   then httpd_output_pause() will not do anything, it
+		   will not fill the buffer and it will not update the
+		   timer; therefore, we reset the timer here */
+		timer_reset(httpd->timer);
+
+		/* some arbitrary delay that is long enough to avoid
+		   consuming too much CPU, and short enough to notice
+		   new clients quickly enough */
+		return 1000;
+	}
+
+	return httpd->timer->started
+		? timer_delay(httpd->timer)
+		: 0;
+}
+
+void
+HttpdOutput::BroadcastPage(Page *page)
+{
+	assert(page != NULL);
+
+	const ScopeLock protect(mutex);
+	for (auto &client : clients)
+		client.PushPage(page);
+}
+
+void
+HttpdOutput::BroadcastFromEncoder()
+{
+	mutex.lock();
+	for (auto &client : clients) {
+		if (client.GetQueueSize() > 256 * 1024) {
+			g_debug("client is too slow, flushing its queue");
+			client.CancelQueue();
+		}
+	}
+	mutex.unlock();
+
+	Page *page;
+	while ((page = ReadPage()) != nullptr) {
+		BroadcastPage(page);
+		page->Unref();
+	}
+}
+
+inline bool
+HttpdOutput::EncodeAndPlay(const void *chunk, size_t size, GError **error_r)
+{
+	if (!encoder_write(encoder, chunk, size, error_r))
+		return false;
+
+	unflushed_input += size;
+
+	BroadcastFromEncoder();
+	return true;
+}
+
+static size_t
+httpd_output_play(struct audio_output *ao, const void *chunk, size_t size,
+		  GError **error_r)
+{
+	HttpdOutput *httpd = Cast(ao);
+
+	if (httpd->LockHasClients()) {
+		if (!httpd->EncodeAndPlay(chunk, size, error_r))
+			return 0;
+	}
+
+	if (!httpd->timer->started)
+		timer_start(httpd->timer);
+	timer_add(httpd->timer, size);
+
+	return size;
+}
+
+static bool
+httpd_output_pause(struct audio_output *ao)
+{
+	HttpdOutput *httpd = Cast(ao);
+
+	if (httpd->LockHasClients()) {
+		static const char silence[1020] = { 0 };
+		return httpd_output_play(ao, silence, sizeof(silence),
+					 NULL) > 0;
+	} else {
+		return true;
+	}
+}
+
+inline void
+HttpdOutput::SendTag(const struct tag *tag)
+{
+	assert(tag != NULL);
+
+	if (encoder->plugin->tag != NULL) {
+		/* embed encoder tags */
+
+		/* flush the current stream, and end it */
+
+		encoder_pre_tag(encoder, NULL);
+		BroadcastFromEncoder();
+
+		/* send the tag to the encoder - which starts a new
+		   stream now */
+
+		encoder_tag(encoder, tag, NULL);
+
+		/* the first page generated by the encoder will now be
+		   used as the new "header" page, which is sent to all
+		   new clients */
+
+		Page *page = ReadPage();
+		if (page != NULL) {
+			if (header != NULL)
+				header->Unref();
+			header = page;
+			BroadcastPage(page);
+		}
+	} else {
+		/* use Icy-Metadata */
+
+		if (metadata != NULL)
+			metadata->Unref();
+
+		static constexpr tag_type types[] = {
+			TAG_ALBUM, TAG_ARTIST, TAG_TITLE,
+			TAG_NUM_OF_ITEM_TYPES
+		};
+
+		metadata = icy_server_metadata_page(tag, &types[0]);
+		if (metadata != NULL) {
+			const ScopeLock protect(mutex);
+			for (auto &client : clients)
+				client.PushMetaData(metadata);
+		}
+	}
+}
+
+static void
+httpd_output_tag(struct audio_output *ao, const struct tag *tag)
+{
+	HttpdOutput *httpd = Cast(ao);
+
+	httpd->SendTag(tag);
+}
+
+static void
+httpd_output_cancel(struct audio_output *ao)
+{
+	HttpdOutput *httpd = Cast(ao);
+
+	const ScopeLock protect(httpd->mutex);
+	for (auto &client : httpd->clients)
+		client.CancelQueue();
+}
+
+const struct audio_output_plugin httpd_output_plugin = {
+	"httpd",
+	nullptr,
+	httpd_output_init,
+	httpd_output_finish,
+	httpd_output_enable,
+	httpd_output_disable,
+	httpd_output_open,
+	httpd_output_close,
+	httpd_output_delay,
+	httpd_output_tag,
+	httpd_output_play,
+	nullptr,
+	httpd_output_cancel,
+	httpd_output_pause,
+	nullptr,
+};