1 files changed, 261 insertions, 0 deletions
diff --git a/src/decoder/SndfileDecoderPlugin.cxx b/src/decoder/SndfileDecoderPlugin.cxx
new file mode 100644
index 000000000..d884ca2d1
--- /dev/null
+++ b/src/decoder/SndfileDecoderPlugin.cxx
@@ -0,0 +1,261 @@
+/*
+ * 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 "SndfileDecoderPlugin.hxx"
+#include "decoder_api.h"
+#include "audio_check.h"
+#include "tag_handler.h"
+
+#include <sndfile.h>
+
+#undef G_LOG_DOMAIN
+#define G_LOG_DOMAIN "sndfile"
+
+static sf_count_t
+sndfile_vio_get_filelen(void *user_data)
+{
+	const struct input_stream *is = (const struct input_stream *)user_data;
+
+	return input_stream_get_size(is);
+}
+
+static sf_count_t
+sndfile_vio_seek(sf_count_t offset, int whence, void *user_data)
+{
+	struct input_stream *is = (struct input_stream *)user_data;
+	bool success;
+
+	success = input_stream_lock_seek(is, offset, whence, nullptr);
+	if (!success)
+		return -1;
+
+	return input_stream_get_offset(is);
+}
+
+static sf_count_t
+sndfile_vio_read(void *ptr, sf_count_t count, void *user_data)
+{
+	struct input_stream *is = (struct input_stream *)user_data;
+	GError *error = nullptr;
+	size_t nbytes;
+
+	nbytes = input_stream_lock_read(is, ptr, count, &error);
+	if (nbytes == 0 && error != nullptr) {
+		g_warning("%s", error->message);
+		g_error_free(error);
+		return -1;
+	}
+
+	return nbytes;
+}
+
+static sf_count_t
+sndfile_vio_write(G_GNUC_UNUSED const void *ptr,
+		  G_GNUC_UNUSED sf_count_t count,
+		  G_GNUC_UNUSED void *user_data)
+{
+	/* no writing! */
+	return -1;
+}
+
+static sf_count_t
+sndfile_vio_tell(void *user_data)
+{
+	const struct input_stream *is = (const struct input_stream *)user_data;
+
+	return input_stream_get_offset(is);
+}
+
+/**
+ * This SF_VIRTUAL_IO implementation wraps MPD's #input_stream to a
+ * libsndfile stream.
+ */
+static SF_VIRTUAL_IO vio = {
+	sndfile_vio_get_filelen,
+	sndfile_vio_seek,
+	sndfile_vio_read,
+	sndfile_vio_write,
+	sndfile_vio_tell,
+};
+
+/**
+ * Converts a frame number to a timestamp (in seconds).
+ */
+static float
+frame_to_time(sf_count_t frame, const struct audio_format *audio_format)
+{
+	return (float)frame / (float)audio_format->sample_rate;
+}
+
+/**
+ * Converts a timestamp (in seconds) to a frame number.
+ */
+static sf_count_t
+time_to_frame(float t, const struct audio_format *audio_format)
+{
+	return (sf_count_t)(t * audio_format->sample_rate);
+}
+
+static void
+sndfile_stream_decode(struct decoder *decoder, struct input_stream *is)
+{
+	GError *error = nullptr;
+	SNDFILE *sf;
+	SF_INFO info;
+	struct audio_format audio_format;
+	size_t frame_size;
+	sf_count_t read_frames, num_frames;
+	int buffer[4096];
+	enum decoder_command cmd;
+
+	info.format = 0;
+
+	sf = sf_open_virtual(&vio, SFM_READ, &info, is);
+	if (sf == nullptr) {
+		g_warning("sf_open_virtual() failed");
+		return;
+	}
+
+	/* for now, always read 32 bit samples.  Later, we could lower
+	   MPD's CPU usage by reading 16 bit samples with
+	   sf_readf_short() on low-quality source files. */
+	if (!audio_format_init_checked(&audio_format, info.samplerate,
+				       SAMPLE_FORMAT_S32,
+				       info.channels, &error)) {
+		g_warning("%s", error->message);
+		g_error_free(error);
+		return;
+	}
+
+	decoder_initialized(decoder, &audio_format, info.seekable,
+			    frame_to_time(info.frames, &audio_format));
+
+	frame_size = audio_format_frame_size(&audio_format);
+	read_frames = sizeof(buffer) / frame_size;
+
+	do {
+		num_frames = sf_readf_int(sf, buffer, read_frames);
+		if (num_frames <= 0)
+			break;
+
+		cmd = decoder_data(decoder, is,
+				   buffer, num_frames * frame_size,
+				   0);
+		if (cmd == DECODE_COMMAND_SEEK) {
+			sf_count_t c =
+				time_to_frame(decoder_seek_where(decoder),
+					      &audio_format);
+			c = sf_seek(sf, c, SEEK_SET);
+			if (c < 0)
+				decoder_seek_error(decoder);
+			else
+				decoder_command_finished(decoder);
+			cmd = DECODE_COMMAND_NONE;
+		}
+	} while (cmd == DECODE_COMMAND_NONE);
+
+	sf_close(sf);
+}
+
+static bool
+sndfile_scan_file(const char *path_fs,
+		  const struct tag_handler *handler, void *handler_ctx)
+{
+	SNDFILE *sf;
+	SF_INFO info;
+	const char *p;
+
+	info.format = 0;
+
+	sf = sf_open(path_fs, SFM_READ, &info);
+	if (sf == nullptr)
+		return false;
+
+	if (!audio_valid_sample_rate(info.samplerate)) {
+		sf_close(sf);
+		g_warning("Invalid sample rate in %s\n", path_fs);
+		return false;
+	}
+
+	tag_handler_invoke_duration(handler, handler_ctx,
+				    info.frames / info.samplerate);
+
+	p = sf_get_string(sf, SF_STR_TITLE);
+	if (p != nullptr)
+		tag_handler_invoke_tag(handler, handler_ctx,
+				       TAG_TITLE, p);
+
+	p = sf_get_string(sf, SF_STR_ARTIST);
+	if (p != nullptr)
+		tag_handler_invoke_tag(handler, handler_ctx,
+				       TAG_ARTIST, p);
+
+	p = sf_get_string(sf, SF_STR_DATE);
+	if (p != nullptr)
+		tag_handler_invoke_tag(handler, handler_ctx,
+				       TAG_DATE, p);
+
+	sf_close(sf);
+
+	return true;
+}
+
+static const char *const sndfile_suffixes[] = {
+	"wav", "aiff", "aif", /* Microsoft / SGI / Apple */
+	"au", "snd", /* Sun / DEC / NeXT */
+	"paf", /* Paris Audio File */
+	"iff", "svx", /* Commodore Amiga IFF / SVX */
+	"sf", /* IRCAM */
+	"voc", /* Creative */
+	"w64", /* Soundforge */
+	"pvf", /* Portable Voice Format */
+	"xi", /* Fasttracker */
+	"htk", /* HMM Tool Kit */
+	"caf", /* Apple */
+	"sd2", /* Sound Designer II */
+
+	/* libsndfile also supports FLAC and Ogg Vorbis, but only by
+	   linking with libFLAC and libvorbis - we can do better, we
+	   have native plugins for these libraries */
+
+	nullptr
+};
+
+static const char *const sndfile_mime_types[] = {
+	"audio/x-wav",
+	"audio/x-aiff",
+
+	/* what are the MIME types of the other supported formats? */
+
+	nullptr
+};
+
+const struct decoder_plugin sndfile_decoder_plugin = {
+	"sndfile",
+	nullptr,
+	nullptr,
+	sndfile_stream_decode,
+	nullptr,
+	sndfile_scan_file,
+	nullptr,
+	nullptr,
+	sndfile_suffixes,
+	sndfile_mime_types,
+};