From d3641766a580d20c4848d056a78399f10fcc6f18 Mon Sep 17 00:00:00 2001 From: Max Kellermann Date: Sun, 28 Jul 2013 12:42:06 +0200 Subject: decoder/sndfile: convert to C++ --- src/decoder/SndfileDecoderPlugin.cxx | 261 +++++++++++++++++++++++++++++++++++ src/decoder/SndfileDecoderPlugin.hxx | 25 ++++ src/decoder/sndfile_decoder_plugin.c | 255 ---------------------------------- 3 files changed, 286 insertions(+), 255 deletions(-) create mode 100644 src/decoder/SndfileDecoderPlugin.cxx create mode 100644 src/decoder/SndfileDecoderPlugin.hxx delete mode 100644 src/decoder/sndfile_decoder_plugin.c (limited to 'src/decoder') 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 + +#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, +}; diff --git a/src/decoder/SndfileDecoderPlugin.hxx b/src/decoder/SndfileDecoderPlugin.hxx new file mode 100644 index 000000000..ba60fafd0 --- /dev/null +++ b/src/decoder/SndfileDecoderPlugin.hxx @@ -0,0 +1,25 @@ +/* + * 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. + */ + +#ifndef MPD_DECODER_SNDFILE_HXX +#define MPD_DECODER_SNDFILE_HXX + +extern const struct decoder_plugin sndfile_decoder_plugin; + +#endif 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 - -#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, -}; -- cgit v1.2.3