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author | Max Kellermann <max@duempel.org> | 2009-07-07 08:58:51 +0200 |
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committer | Max Kellermann <max@duempel.org> | 2009-07-07 08:58:51 +0200 |
commit | 1eebbc746f715e32f165ed62fdc57447a5903b21 (patch) | |
tree | 95a70858bac7aea6bf6bd59c26a67144abf2f6c6 /src/decoder/sndfile_decoder_plugin.c | |
parent | adb2f66cedcac56eaaaa36e8026b497c96c522e6 (diff) | |
download | mpd-1eebbc746f715e32f165ed62fdc57447a5903b21.tar.gz mpd-1eebbc746f715e32f165ed62fdc57447a5903b21.tar.xz mpd-1eebbc746f715e32f165ed62fdc57447a5903b21.zip |
decoder/sndfile: new decoder plugin based on libsndfile
Diffstat (limited to '')
-rw-r--r-- | src/decoder/sndfile_decoder_plugin.c | 246 |
1 files changed, 246 insertions, 0 deletions
diff --git a/src/decoder/sndfile_decoder_plugin.c b/src/decoder/sndfile_decoder_plugin.c new file mode 100644 index 000000000..0c5d2f063 --- /dev/null +++ b/src/decoder/sndfile_decoder_plugin.c @@ -0,0 +1,246 @@ +/* + * Copyright (C) 2003-2009 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 "decoder_api.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 is->size; +} + +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_seek(is, offset, whence); + if (!success) + return -1; + + return is->offset; +} + +static sf_count_t +sndfile_vio_read(void *ptr, sf_count_t count, void *user_data) +{ + struct input_stream *is = user_data; + size_t nbytes; + + nbytes = input_stream_read(is, ptr, count); + if (nbytes == 0 && is->error != 0) + 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 is->offset; +} + +/** + * 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) +{ + SNDFILE *sf; + SF_INFO info; + struct audio_format audio_format; + size_t frame_size; + sf_count_t read_frames, num_frames, position = 0; + 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; + } + + audio_format.sample_rate = info.samplerate; + /* 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. */ + audio_format.bits = 32; + audio_format.channels = info.channels; + + if (!audio_format_valid(&audio_format)) { + g_warning("invalid audio format"); + 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, + frame_to_time(position, &audio_format), + 0, NULL); + 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 struct tag * +sndfile_tag_dup(const char *path_fs) +{ + SNDFILE *sf; + SF_INFO info; + struct tag *tag; + const char *p; + + info.format = 0; + + sf = sf_open(path_fs, SFM_READ, &info); + if (sf == NULL) + return NULL; + + if (!audio_valid_sample_rate(info.samplerate)) { + sf_close(sf); + g_warning("Invalid sample rate in %s\n", path_fs); + return NULL; + } + + tag = tag_new(); + tag->time = info.frames / info.samplerate; + + p = sf_get_string(sf, SF_STR_TITLE); + if (p != NULL) + tag_add_item(tag, TAG_ITEM_TITLE, p); + + p = sf_get_string(sf, SF_STR_ARTIST); + if (p != NULL) + tag_add_item(tag, TAG_ITEM_ARTIST, p); + + p = sf_get_string(sf, SF_STR_DATE); + if (p != NULL) + tag_add_item(tag, TAG_ITEM_DATE, p); + + sf_close(sf); + + return tag; +} + +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, + .tag_dup = sndfile_tag_dup, + .suffixes = sndfile_suffixes, + .mime_types = sndfile_mime_types, +}; |