decoder/sndfile: convert to C++

1 files changed, 0 insertions, 255 deletions

diff --git a/src/decoder/sndfile_decoder_plugin.c b/src/decoder/sndfile_decoder_plugin.c
deleted file mode 100644
index e70a2dc2e..000000000
--- a/src/decoder/sndfile_decoder_plugin.c
+++ /dev/null
@@ -1,255 +0,0 @@
-/*
- * 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 "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 = 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 = user_data;
-	bool success;
-
-	success = input_stream_lock_seek(is, offset, whence, NULL);
-	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 = user_data;
-	GError *error = NULL;
-	size_t nbytes;
-
-	nbytes = input_stream_lock_read(is, ptr, count, &error);
-	if (nbytes == 0 && error != NULL) {
-		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 = 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 = {
-	.get_filelen = sndfile_vio_get_filelen,
-	.seek = sndfile_vio_seek,
-	.read = sndfile_vio_read,
-	.write = sndfile_vio_write,
-	.tell = 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 = NULL;
-	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 == NULL) {
-		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 == NULL)
-		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 != NULL)
-		tag_handler_invoke_tag(handler, handler_ctx,
-				       TAG_TITLE, p);
-
-	p = sf_get_string(sf, SF_STR_ARTIST);
-	if (p != NULL)
-		tag_handler_invoke_tag(handler, handler_ctx,
-				       TAG_ARTIST, p);
-
-	p = sf_get_string(sf, SF_STR_DATE);
-	if (p != NULL)
-		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 */
-
-	NULL
-};
-
-static const char *const sndfile_mime_types[] = {
-	"audio/x-wav",
-	"audio/x-aiff",
-
-	/* what are the MIME types of the other supported formats? */
-
-	NULL
-};
-
-const struct decoder_plugin sndfile_decoder_plugin = {
-	.name = "sndfile",
-	.stream_decode = sndfile_stream_decode,
-	.scan_file = sndfile_scan_file,
-	.suffixes = sndfile_suffixes,
-	.mime_types = sndfile_mime_types,
-};