1 files changed, 295 insertions, 0 deletions
diff --git a/src/decoder/plugins/SndfileDecoderPlugin.cxx b/src/decoder/plugins/SndfileDecoderPlugin.cxx
new file mode 100644
index 000000000..cf3aa61d5
--- /dev/null
+++ b/src/decoder/plugins/SndfileDecoderPlugin.cxx
@@ -0,0 +1,295 @@
+/*
+ * Copyright (C) 2003-2014 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 "../DecoderAPI.hxx"
+#include "input/InputStream.hxx"
+#include "CheckAudioFormat.hxx"
+#include "tag/TagHandler.hxx"
+#include "fs/Path.hxx"
+#include "util/Error.hxx"
+#include "util/Domain.hxx"
+#include "Log.hxx"
+
+#include <sndfile.h>
+
+static constexpr Domain sndfile_domain("sndfile");
+
+static sf_count_t
+sndfile_vio_get_filelen(void *user_data)
+{
+	const InputStream &is = *(const InputStream *)user_data;
+
+	return is.GetSize();
+}
+
+static sf_count_t
+sndfile_vio_seek(sf_count_t _offset, int whence, void *user_data)
+{
+	InputStream &is = *(InputStream *)user_data;
+
+	InputStream::offset_type offset = _offset;
+	switch (whence) {
+	case SEEK_SET:
+		break;
+
+	case SEEK_CUR:
+		offset += is.GetOffset();
+		break;
+
+	case SEEK_END:
+		if (!is.KnownSize())
+			return -1;
+
+		offset += is.GetSize();
+		break;
+
+	default:
+		return -1;
+	}
+
+	if (!is.LockSeek(offset, IgnoreError()))
+		return -1;
+
+	return is.GetOffset();
+}
+
+static sf_count_t
+sndfile_vio_read(void *ptr, sf_count_t count, void *user_data)
+{
+	InputStream &is = *(InputStream *)user_data;
+
+	sf_count_t total_bytes = 0;
+	Error error;
+
+	/* this loop is necessary because libsndfile chokes on partial
+	   reads */
+
+	do {
+		size_t nbytes = is.LockRead((char *)ptr + total_bytes,
+					    count - total_bytes, error);
+		if (nbytes == 0) {
+			if (error.IsDefined()) {
+				LogError(error);
+				return -1;
+			}
+
+			break;
+		}
+
+		total_bytes += nbytes;
+	} while (total_bytes < count);
+
+	return total_bytes;
+}
+
+static sf_count_t
+sndfile_vio_write(gcc_unused const void *ptr,
+		  gcc_unused sf_count_t count,
+		  gcc_unused void *user_data)
+{
+	/* no writing! */
+	return -1;
+}
+
+static sf_count_t
+sndfile_vio_tell(void *user_data)
+{
+	const InputStream &is = *(const InputStream *)user_data;
+
+	return is.GetOffset();
+}
+
+/**
+ * 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 AudioFormat *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 AudioFormat *audio_format)
+{
+	return (sf_count_t)(t * audio_format->sample_rate);
+}
+
+static void
+sndfile_stream_decode(Decoder &decoder, InputStream &is)
+{
+	SNDFILE *sf;
+	SF_INFO info;
+	size_t frame_size;
+	sf_count_t read_frames, num_frames;
+	int buffer[4096];
+
+	info.format = 0;
+
+	sf = sf_open_virtual(&vio, SFM_READ, &info, &is);
+	if (sf == nullptr) {
+		LogWarning(sndfile_domain, "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. */
+	Error error;
+	AudioFormat audio_format;
+	if (!audio_format_init_checked(audio_format, info.samplerate,
+				       SampleFormat::S32,
+				       info.channels, error)) {
+		LogError(error);
+		return;
+	}
+
+	decoder_initialized(decoder, audio_format, info.seekable,
+			    frame_to_time(info.frames, &audio_format));
+
+	frame_size = audio_format.GetFrameSize();
+	read_frames = sizeof(buffer) / frame_size;
+
+	DecoderCommand cmd;
+	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 == DecoderCommand::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 = DecoderCommand::NONE;
+		}
+	} while (cmd == DecoderCommand::NONE);
+
+	sf_close(sf);
+}
+
+static bool
+sndfile_scan_file(Path 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.c_str(), SFM_READ, &info);
+	if (sf == nullptr)
+		return false;
+
+	if (!audio_valid_sample_rate(info.samplerate)) {
+		sf_close(sf);
+		FormatWarning(sndfile_domain,
+			      "Invalid sample rate in %s", path_fs.c_str());
+		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 DecoderPlugin sndfile_decoder_plugin = {
+	"sndfile",
+	nullptr,
+	nullptr,
+	sndfile_stream_decode,
+	nullptr,
+	sndfile_scan_file,
+	nullptr,
+	nullptr,
+	sndfile_suffixes,
+	sndfile_mime_types,
+};