diff options
Diffstat (limited to 'src/decoder')
| -rw-r--r-- | src/decoder/_flac_common.c | 431 | ||||
| -rw-r--r-- | src/decoder/_flac_common.h | 183 | ||||
| -rw-r--r-- | src/decoder/_ogg_common.c | 4 | ||||
| -rw-r--r-- | src/decoder/_ogg_common.h | 4 | ||||
| -rw-r--r-- | src/decoder/audiofile_decoder_plugin.c (renamed from src/decoder/audiofile_plugin.c) | 103 | ||||
| -rw-r--r-- | src/decoder/faad_decoder_plugin.c (renamed from src/decoder/faad_plugin.c) | 96 | ||||
| -rw-r--r-- | src/decoder/ffmpeg_decoder_plugin.c (renamed from src/decoder/ffmpeg_plugin.c) | 469 | ||||
| -rw-r--r-- | src/decoder/flac_compat.h | 114 | ||||
| -rw-r--r-- | src/decoder/flac_decoder_plugin.c | 497 | ||||
| -rw-r--r-- | src/decoder/flac_metadata.c | 286 | ||||
| -rw-r--r-- | src/decoder/flac_metadata.h | 55 | ||||
| -rw-r--r-- | src/decoder/flac_pcm.c | 109 | ||||
| -rw-r--r-- | src/decoder/flac_pcm.h | 33 | ||||
| -rw-r--r-- | src/decoder/flac_plugin.c | 918 | ||||
| -rw-r--r-- | src/decoder/fluidsynth_decoder_plugin.c (renamed from src/decoder/fluidsynth_plugin.c) | 13 | ||||
| -rw-r--r-- | src/decoder/gme_decoder_plugin.c | 247 | ||||
| -rw-r--r-- | src/decoder/mad_decoder_plugin.c (renamed from src/decoder/mad_plugin.c) | 292 | ||||
| -rw-r--r-- | src/decoder/mikmod_decoder_plugin.c (renamed from src/decoder/mikmod_plugin.c) | 170 | ||||
| -rw-r--r-- | src/decoder/modplug_decoder_plugin.c (renamed from src/decoder/modplug_plugin.c) | 67 | ||||
| -rw-r--r-- | src/decoder/mp4ff_decoder_plugin.c (renamed from src/decoder/mp4ff_plugin.c) | 124 | ||||
| -rw-r--r-- | src/decoder/mpcdec_decoder_plugin.c (renamed from src/decoder/mpcdec_plugin.c) | 93 | ||||
| -rw-r--r-- | src/decoder/mpg123_decoder_plugin.c | 209 | ||||
| -rw-r--r-- | src/decoder/oggflac_decoder_plugin.c (renamed from src/decoder/oggflac_plugin.c) | 84 | ||||
| -rw-r--r-- | src/decoder/sidplay_decoder_plugin.cxx | 429 | ||||
| -rw-r--r-- | src/decoder/sidplay_plugin.cxx | 163 | ||||
| -rw-r--r-- | src/decoder/sndfile_decoder_plugin.c | 251 | ||||
| -rw-r--r-- | src/decoder/vorbis_decoder_plugin.c (renamed from src/decoder/vorbis_plugin.c) | 260 | ||||
| -rw-r--r-- | src/decoder/wavpack_decoder_plugin.c (renamed from src/decoder/wavpack_plugin.c) | 188 | ||||
| -rw-r--r-- | src/decoder/wildmidi_decoder_plugin.c (renamed from src/decoder/wildmidi_plugin.c) | 22 |
29 files changed, 3342 insertions, 2572 deletions
diff --git a/src/decoder/_flac_common.c b/src/decoder/_flac_common.c index 7c8fe9875..8dd22a253 100644 --- a/src/decoder/_flac_common.c +++ b/src/decoder/_flac_common.c @@ -1,5 +1,5 @@ /* - * Copyright (C) 2003-2009 The Music Player Daemon Project + * Copyright (C) 2003-2010 The Music Player Daemon Project * http://www.musicpd.org * * This program is free software; you can redistribute it and/or modify @@ -21,7 +21,11 @@ * Common data structures and functions used by FLAC and OggFLAC */ +#include "config.h" #include "_flac_common.h" +#include "flac_metadata.h" +#include "flac_pcm.h" +#include "audio_check.h" #include <glib.h> @@ -31,186 +35,104 @@ void flac_data_init(struct flac_data *data, struct decoder * decoder, struct input_stream *input_stream) { - data->time = 0; + pcm_buffer_init(&data->buffer); + + data->unsupported = false; + data->initialized = false; + data->total_frames = 0; + data->first_frame = 0; + data->next_frame = 0; + data->position = 0; - data->bit_rate = 0; data->decoder = decoder; data->input_stream = input_stream; - data->replay_gain_info = NULL; data->tag = NULL; } -static void -flac_find_float_comment(const FLAC__StreamMetadata *block, - const char *cmnt, float *fl, bool *found_r) +void +flac_data_deinit(struct flac_data *data) { - int offset; - size_t pos; - int len; - unsigned char tmp, *p; - - offset = FLAC__metadata_object_vorbiscomment_find_entry_from(block, 0, - cmnt); - if (offset < 0) - return; - - pos = strlen(cmnt) + 1; /* 1 is for '=' */ - len = block->data.vorbis_comment.comments[offset].length - pos; - if (len <= 0) - return; - - p = &block->data.vorbis_comment.comments[offset].entry[pos]; - tmp = p[len]; - p[len] = '\0'; - *fl = (float)atof((char *)p); - p[len] = tmp; - - *found_r = true; -} + pcm_buffer_deinit(&data->buffer); -static void -flac_parse_replay_gain(const FLAC__StreamMetadata *block, - struct flac_data *data) -{ - bool found = false; - - if (data->replay_gain_info) - replay_gain_info_free(data->replay_gain_info); - - data->replay_gain_info = replay_gain_info_new(); - - flac_find_float_comment(block, "replaygain_album_gain", - &data->replay_gain_info->tuples[REPLAY_GAIN_ALBUM].gain, - &found); - flac_find_float_comment(block, "replaygain_album_peak", - &data->replay_gain_info->tuples[REPLAY_GAIN_ALBUM].peak, - &found); - flac_find_float_comment(block, "replaygain_track_gain", - &data->replay_gain_info->tuples[REPLAY_GAIN_TRACK].gain, - &found); - flac_find_float_comment(block, "replaygain_track_peak", - &data->replay_gain_info->tuples[REPLAY_GAIN_TRACK].peak, - &found); - - if (!found) { - replay_gain_info_free(data->replay_gain_info); - data->replay_gain_info = NULL; - } + if (data->tag != NULL) + tag_free(data->tag); } -/** - * Checks if the specified name matches the entry's name, and if yes, - * returns the comment value (not null-temrinated). - */ -static const char * -flac_comment_value(const FLAC__StreamMetadata_VorbisComment_Entry *entry, - const char *name, const char *char_tnum, size_t *length_r) +static enum sample_format +flac_sample_format(unsigned bits_per_sample) { - size_t name_length = strlen(name); - size_t char_tnum_length = 0; - const char *comment = (const char*)entry->entry; - - if (entry->length <= name_length || - g_ascii_strncasecmp(comment, name, name_length) != 0) - return NULL; - - if (char_tnum != NULL) { - char_tnum_length = strlen(char_tnum); - if (entry->length > name_length + char_tnum_length + 2 && - comment[name_length] == '[' && - g_ascii_strncasecmp(comment + name_length + 1, - char_tnum, char_tnum_length) == 0 && - comment[name_length + char_tnum_length + 1] == ']') - name_length = name_length + char_tnum_length + 2; - else if (entry->length > name_length + char_tnum_length && - g_ascii_strncasecmp(comment + name_length, - char_tnum, char_tnum_length) == 0) - name_length = name_length + char_tnum_length; - } + switch (bits_per_sample) { + case 8: + return SAMPLE_FORMAT_S8; - if (comment[name_length] == '=') { - *length_r = entry->length - name_length - 1; - return comment + name_length + 1; - } + case 16: + return SAMPLE_FORMAT_S16; - return NULL; -} + case 24: + return SAMPLE_FORMAT_S24_P32; -/** - * Check if the comment's name equals the passed name, and if so, copy - * the comment value into the tag. - */ -static bool -flac_copy_comment(struct tag *tag, - const FLAC__StreamMetadata_VorbisComment_Entry *entry, - const char *name, enum tag_type tag_type, - const char *char_tnum) -{ - const char *value; - size_t value_length; + case 32: + return SAMPLE_FORMAT_S32; - value = flac_comment_value(entry, name, char_tnum, &value_length); - if (value != NULL) { - tag_add_item_n(tag, tag_type, value, value_length); - return true; + default: + return SAMPLE_FORMAT_UNDEFINED; } - - return false; } -/* tracknumber is used in VCs, MPD uses "track" ..., all the other - * tag names match */ -static const char *VORBIS_COMMENT_TRACK_KEY = "tracknumber"; -static const char *VORBIS_COMMENT_DISC_KEY = "discnumber"; - static void -flac_parse_comment(struct tag *tag, const char *char_tnum, - const FLAC__StreamMetadata_VorbisComment_Entry *entry) +flac_got_stream_info(struct flac_data *data, + const FLAC__StreamMetadata_StreamInfo *stream_info) { - assert(tag != NULL); - - if (flac_copy_comment(tag, entry, VORBIS_COMMENT_TRACK_KEY, - TAG_ITEM_TRACK, char_tnum) || - flac_copy_comment(tag, entry, VORBIS_COMMENT_DISC_KEY, - TAG_ITEM_DISC, char_tnum) || - flac_copy_comment(tag, entry, "album artist", - TAG_ITEM_ALBUM_ARTIST, char_tnum)) + if (data->initialized || data->unsupported) return; - for (unsigned i = 0; i < TAG_NUM_OF_ITEM_TYPES; ++i) - if (flac_copy_comment(tag, entry, - tag_item_names[i], i, char_tnum)) - return; -} + GError *error = NULL; + if (!audio_format_init_checked(&data->audio_format, + stream_info->sample_rate, + flac_sample_format(stream_info->bits_per_sample), + stream_info->channels, &error)) { + g_warning("%s", error->message); + g_error_free(error); + data->unsupported = true; + return; + } -void -flac_vorbis_comments_to_tag(struct tag *tag, const char *char_tnum, - const FLAC__StreamMetadata *block) -{ - FLAC__StreamMetadata_VorbisComment_Entry *comments = - block->data.vorbis_comment.comments; + data->frame_size = audio_format_frame_size(&data->audio_format); - for (unsigned i = block->data.vorbis_comment.num_comments; i > 0; --i) - flac_parse_comment(tag, char_tnum, comments++); + if (data->total_frames == 0) + data->total_frames = stream_info->total_samples; + + data->initialized = true; } void flac_metadata_common_cb(const FLAC__StreamMetadata * block, struct flac_data *data) { - const FLAC__StreamMetadata_StreamInfo *si = &(block->data.stream_info); + if (data->unsupported) + return; + + struct replay_gain_info rgi; + char *mixramp_start; + char *mixramp_end; + float replay_gain_db = 0; switch (block->type) { case FLAC__METADATA_TYPE_STREAMINFO: - data->audio_format.bits = (int8_t)si->bits_per_sample; - data->audio_format.sample_rate = si->sample_rate; - data->audio_format.channels = (int8_t)si->channels; - data->total_time = ((float)si->total_samples) / (si->sample_rate); + flac_got_stream_info(data, &block->data.stream_info); break; + case FLAC__METADATA_TYPE_VORBIS_COMMENT: - flac_parse_replay_gain(block, data); + if (flac_parse_replay_gain(&rgi, block)) + replay_gain_db = decoder_replay_gain(data->decoder, &rgi); + if (flac_parse_mixramp(&mixramp_start, &mixramp_end, block)) { + g_debug("setting mixramp_tags"); + decoder_mixramp(data->decoder, replay_gain_db, + mixramp_start, mixramp_end); + } if (data->tag != NULL) - flac_vorbis_comments_to_tag(data->tag, NULL, block); + flac_vorbis_comments_to_tag(data->tag, NULL, + &block->data.vorbis_comment); default: break; @@ -239,187 +161,82 @@ void flac_error_common_cb(const char *plugin, } } -static void flac_convert_stereo16(int16_t *dest, - const FLAC__int32 * const buf[], - unsigned int position, unsigned int end) -{ - for (; position < end; ++position) { - *dest++ = buf[0][position]; - *dest++ = buf[1][position]; - } -} - -static void -flac_convert_16(int16_t *dest, - unsigned int num_channels, - const FLAC__int32 * const buf[], - unsigned int position, unsigned int end) -{ - unsigned int c_chan; - - for (; position < end; ++position) - for (c_chan = 0; c_chan < num_channels; c_chan++) - *dest++ = buf[c_chan][position]; -} - /** - * Note: this function also handles 24 bit files! + * This function attempts to call decoder_initialized() in case there + * was no STREAMINFO block. This is allowed for nonseekable streams, + * where the server sends us only a part of the file, without + * providing the STREAMINFO block from the beginning of the file + * (e.g. when seeking with SqueezeBox Server). */ -static void -flac_convert_32(int32_t *dest, - unsigned int num_channels, - const FLAC__int32 * const buf[], - unsigned int position, unsigned int end) -{ - unsigned int c_chan; - - for (; position < end; ++position) - for (c_chan = 0; c_chan < num_channels; c_chan++) - *dest++ = buf[c_chan][position]; -} - -static void -flac_convert_8(int8_t *dest, - unsigned int num_channels, - const FLAC__int32 * const buf[], - unsigned int position, unsigned int end) -{ - unsigned int c_chan; +static bool +flac_got_first_frame(struct flac_data *data, const FLAC__FrameHeader *header) +{ + if (data->unsupported) + return false; + + GError *error = NULL; + if (!audio_format_init_checked(&data->audio_format, + header->sample_rate, + flac_sample_format(header->bits_per_sample), + header->channels, &error)) { + g_warning("%s", error->message); + g_error_free(error); + data->unsupported = true; + return false; + } - for (; position < end; ++position) - for (c_chan = 0; c_chan < num_channels; c_chan++) - *dest++ = buf[c_chan][position]; -} + data->frame_size = audio_format_frame_size(&data->audio_format); -static void flac_convert(unsigned char *dest, - unsigned int num_channels, - unsigned int bytes_per_sample, - const FLAC__int32 * const buf[], - unsigned int position, unsigned int end) -{ - switch (bytes_per_sample) { - case 2: - if (num_channels == 2) - flac_convert_stereo16((int16_t*)dest, buf, - position, end); - else - flac_convert_16((int16_t*)dest, num_channels, buf, - position, end); - break; + decoder_initialized(data->decoder, &data->audio_format, + data->input_stream->seekable, + (float)data->total_frames / + (float)data->audio_format.sample_rate); - case 4: - flac_convert_32((int32_t*)dest, num_channels, buf, - position, end); - break; + data->initialized = true; - case 1: - flac_convert_8((int8_t*)dest, num_channels, buf, - position, end); - break; - } + return true; } FLAC__StreamDecoderWriteStatus flac_common_write(struct flac_data *data, const FLAC__Frame * frame, - const FLAC__int32 *const buf[]) + const FLAC__int32 *const buf[], + FLAC__uint64 nbytes) { - unsigned int c_samp; - const unsigned int num_channels = frame->header.channels; - const unsigned int bytes_per_sample = - audio_format_sample_size(&data->audio_format); - const unsigned int bytes_per_channel = - bytes_per_sample * frame->header.channels; - const unsigned int max_samples = FLAC_CHUNK_SIZE / bytes_per_channel; - unsigned int num_samples; enum decoder_command cmd; + void *buffer; + unsigned bit_rate; - if (bytes_per_sample != 1 && bytes_per_sample != 2 && - bytes_per_sample != 4) - /* exotic unsupported bit rate */ + if (!data->initialized && !flac_got_first_frame(data, &frame->header)) return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT; - for (c_samp = 0; c_samp < frame->header.blocksize; - c_samp += num_samples) { - num_samples = frame->header.blocksize - c_samp; - if (num_samples > max_samples) - num_samples = max_samples; - - flac_convert(data->chunk, - num_channels, bytes_per_sample, buf, - c_samp, c_samp + num_samples); - - cmd = decoder_data(data->decoder, data->input_stream, - data->chunk, - num_samples * bytes_per_channel, - data->time, data->bit_rate, - data->replay_gain_info); - switch (cmd) { - case DECODE_COMMAND_NONE: - case DECODE_COMMAND_START: - break; - - case DECODE_COMMAND_STOP: - return - FLAC__STREAM_DECODER_WRITE_STATUS_ABORT; - - case DECODE_COMMAND_SEEK: - return - FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE; - } - } - - return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE; -} - -#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7 + size_t buffer_size = frame->header.blocksize * data->frame_size; + buffer = pcm_buffer_get(&data->buffer, buffer_size); -char* -flac_cue_track( const char* pathname, - const unsigned int tnum) -{ - FLAC__bool success; - FLAC__StreamMetadata* cs; - - success = FLAC__metadata_get_cuesheet(pathname, &cs); - if (!success) - return NULL; - - assert(cs != NULL); - - if (cs->data.cue_sheet.num_tracks <= 1) - { - FLAC__metadata_object_delete(cs); - return NULL; - } + flac_convert(buffer, frame->header.channels, + data->audio_format.format, buf, + 0, frame->header.blocksize); - if (tnum > 0 && tnum < cs->data.cue_sheet.num_tracks) - { - char* track = g_strdup_printf("track_%03u.flac", tnum); + if (nbytes > 0) + bit_rate = nbytes * 8 * frame->header.sample_rate / + (1000 * frame->header.blocksize); + else + bit_rate = 0; + + cmd = decoder_data(data->decoder, data->input_stream, + buffer, buffer_size, + bit_rate); + data->next_frame += frame->header.blocksize; + switch (cmd) { + case DECODE_COMMAND_NONE: + case DECODE_COMMAND_START: + break; - FLAC__metadata_object_delete(cs); + case DECODE_COMMAND_STOP: + return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT; - return track; - } - else - { - FLAC__metadata_object_delete(cs); - return NULL; + case DECODE_COMMAND_SEEK: + return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE; } -} -unsigned int -flac_vtrack_tnum(const char* fname) -{ - /* find last occurrence of '_' in fname - * which is hopefully something like track_xxx.flac - * another/better way would be to use tag struct - */ - char* ptr = strrchr(fname, '_'); - if (ptr == NULL) - return 0; - - // copy ascii tracknumber to int - return (unsigned int)strtol(++ptr, NULL, 10); + return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE; } - -#endif /* FLAC_API_VERSION_CURRENT >= 7 */ diff --git a/src/decoder/_flac_common.h b/src/decoder/_flac_common.h index 68de7e969..5c59ee123 100644 --- a/src/decoder/_flac_common.h +++ b/src/decoder/_flac_common.h @@ -1,5 +1,5 @@ /* - * Copyright (C) 2003-2009 The Music Player Daemon Project + * Copyright (C) 2003-2010 The Music Player Daemon Project * http://www.musicpd.org * * This program is free software; you can redistribute it and/or modify @@ -24,136 +24,62 @@ #ifndef MPD_FLAC_COMMON_H #define MPD_FLAC_COMMON_H -#include "../decoder_api.h" -#include "config.h" +#include "decoder_api.h" +#include "pcm_buffer.h" #include <glib.h> +#include <FLAC/stream_decoder.h> +#include <FLAC/metadata.h> + #undef G_LOG_DOMAIN #define G_LOG_DOMAIN "flac" -#include <FLAC/export.h> -#if !defined(FLAC_API_VERSION_CURRENT) || FLAC_API_VERSION_CURRENT <= 7 -# include <FLAC/seekable_stream_decoder.h> -# define flac_decoder FLAC__SeekableStreamDecoder -# define flac_new() FLAC__seekable_stream_decoder_new() - -# define flac_ogg_init(a,b,c,d,e,f,g,h,i,j) (0) - -# define flac_get_decode_position(x,y) \ - FLAC__seekable_stream_decoder_get_decode_position(x,y) -# define flac_get_state(x) FLAC__seekable_stream_decoder_get_state(x) -# define flac_process_single(x) FLAC__seekable_stream_decoder_process_single(x) -# define flac_process_metadata(x) \ - FLAC__seekable_stream_decoder_process_until_end_of_metadata(x) -# define flac_seek_absolute(x,y) \ - FLAC__seekable_stream_decoder_seek_absolute(x,y) -# define flac_finish(x) FLAC__seekable_stream_decoder_finish(x) -# define flac_delete(x) FLAC__seekable_stream_decoder_delete(x) - -# define flac_decoder_eof FLAC__SEEKABLE_STREAM_DECODER_END_OF_STREAM - -typedef unsigned flac_read_status_size_t; -# define flac_read_status FLAC__SeekableStreamDecoderReadStatus -# define flac_read_status_continue \ - FLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_OK -# define flac_read_status_eof FLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_OK -# define flac_read_status_abort \ - FLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_ERROR - -# define flac_seek_status FLAC__SeekableStreamDecoderSeekStatus -# define flac_seek_status_ok FLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_OK -# define flac_seek_status_error FLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_ERROR - -# define flac_tell_status FLAC__SeekableStreamDecoderTellStatus -# define flac_tell_status_ok FLAC__SEEKABLE_STREAM_DECODER_TELL_STATUS_OK -# define flac_tell_status_error \ - FLAC__SEEKABLE_STREAM_DECODER_TELL_STATUS_ERROR -# define flac_tell_status_unsupported \ - FLAC__SEEKABLE_STREAM_DECODER_TELL_STATUS_ERROR - -# define flac_length_status FLAC__SeekableStreamDecoderLengthStatus -# define flac_length_status_ok FLAC__SEEKABLE_STREAM_DECODER_LENGTH_STATUS_OK -# define flac_length_status_error \ - FLAC__SEEKABLE_STREAM_DECODER_LENGTH_STATUS_ERROR -# define flac_length_status_unsupported \ - FLAC__SEEKABLE_STREAM_DECODER_LENGTH_STATUS_ERROR - -# ifdef HAVE_OGGFLAC -# include <OggFLAC/seekable_stream_decoder.h> -# endif -#else /* FLAC_API_VERSION_CURRENT > 7 */ +struct flac_data { + struct pcm_buffer buffer; + + /** + * The size of one frame in the output buffer. + */ + unsigned frame_size; + + /** + * Has decoder_initialized() been called yet? + */ + bool initialized; + + /** + * Does the FLAC file contain an unsupported audio format? + */ + bool unsupported; + + /** + * The validated audio format of the FLAC file. This + * attribute is defined if "initialized" is true. + */ + struct audio_format audio_format; -/* - * OggFLAC support is handled by our flac_plugin already, and - * thus we *can* always have it if libFLAC was compiled with it - */ -# include "_ogg_common.h" - -# include <FLAC/stream_decoder.h> -# define flac_decoder FLAC__StreamDecoder -# define flac_new() FLAC__stream_decoder_new() - -# define flac_init(a,b,c,d,e,f,g,h,i,j) \ - (FLAC__stream_decoder_init_stream(a,b,c,d,e,f,g,h,i,j) \ - == FLAC__STREAM_DECODER_INIT_STATUS_OK) -# define flac_ogg_init(a,b,c,d,e,f,g,h,i,j) \ - (FLAC__stream_decoder_init_ogg_stream(a,b,c,d,e,f,g,h,i,j) \ - == FLAC__STREAM_DECODER_INIT_STATUS_OK) - -# define flac_get_decode_position(x,y) \ - FLAC__stream_decoder_get_decode_position(x,y) -# define flac_get_state(x) FLAC__stream_decoder_get_state(x) -# define flac_process_single(x) FLAC__stream_decoder_process_single(x) -# define flac_process_metadata(x) \ - FLAC__stream_decoder_process_until_end_of_metadata(x) -# define flac_seek_absolute(x,y) FLAC__stream_decoder_seek_absolute(x,y) -# define flac_finish(x) FLAC__stream_decoder_finish(x) -# define flac_delete(x) FLAC__stream_decoder_delete(x) - -# define flac_decoder_eof FLAC__STREAM_DECODER_END_OF_STREAM - -typedef size_t flac_read_status_size_t; -# define flac_read_status FLAC__StreamDecoderReadStatus -# define flac_read_status_continue \ - FLAC__STREAM_DECODER_READ_STATUS_CONTINUE -# define flac_read_status_eof FLAC__STREAM_DECODER_READ_STATUS_END_OF_STREAM -# define flac_read_status_abort FLAC__STREAM_DECODER_READ_STATUS_ABORT - -# define flac_seek_status FLAC__StreamDecoderSeekStatus -# define flac_seek_status_ok FLAC__STREAM_DECODER_SEEK_STATUS_OK -# define flac_seek_status_error FLAC__STREAM_DECODER_SEEK_STATUS_ERROR -# define flac_seek_status_unsupported \ - FLAC__STREAM_DECODER_SEEK_STATUS_UNSUPPORTED - -# define flac_tell_status FLAC__StreamDecoderTellStatus -# define flac_tell_status_ok FLAC__STREAM_DECODER_TELL_STATUS_OK -# define flac_tell_status_error FLAC__STREAM_DECODER_TELL_STATUS_ERROR -# define flac_tell_status_unsupported \ - FLAC__STREAM_DECODER_TELL_STATUS_UNSUPPORTED - -# define flac_length_status FLAC__StreamDecoderLengthStatus -# define flac_length_status_ok FLAC__STREAM_DECODER_LENGTH_STATUS_OK -# define flac_length_status_error FLAC__STREAM_DECODER_LENGTH_STATUS_ERROR -# define flac_length_status_unsupported \ - FLAC__STREAM_DECODER_LENGTH_STATUS_UNSUPPORTED - -#endif /* FLAC_API_VERSION_CURRENT >= 7 */ + /** + * The total number of frames in this song. The decoder + * plugin may initialize this attribute to override the value + * provided by libFLAC (e.g. for sub songs from a CUE sheet). + */ + FLAC__uint64 total_frames; -#include <FLAC/metadata.h> + /** + * The number of the first frame in this song. This is only + * non-zero if playing sub songs from a CUE sheet. + */ + FLAC__uint64 first_frame; -#define FLAC_CHUNK_SIZE 4080 + /** + * The number of the next frame which is going to be decoded. + */ + FLAC__uint64 next_frame; -struct flac_data { - unsigned char chunk[FLAC_CHUNK_SIZE]; - float time; - unsigned int bit_rate; - struct audio_format audio_format; - float total_time; FLAC__uint64 position; struct decoder *decoder; struct input_stream *input_stream; - struct replay_gain_info *replay_gain_info; struct tag *tag; }; @@ -162,6 +88,9 @@ void flac_data_init(struct flac_data *data, struct decoder * decoder, struct input_stream *input_stream); +void +flac_data_deinit(struct flac_data *data); + void flac_metadata_common_cb(const FLAC__StreamMetadata * block, struct flac_data *data); @@ -169,23 +98,9 @@ void flac_error_common_cb(const char *plugin, FLAC__StreamDecoderErrorStatus status, struct flac_data *data); -void -flac_vorbis_comments_to_tag(struct tag *tag, const char *char_tnum, - const FLAC__StreamMetadata *block); - FLAC__StreamDecoderWriteStatus flac_common_write(struct flac_data *data, const FLAC__Frame * frame, - const FLAC__int32 *const buf[]); - -#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7 - -char* -flac_cue_track( const char* pathname, - const unsigned int tnum); - -unsigned int -flac_vtrack_tnum( const char*); - -#endif /* FLAC_API_VERSION_CURRENT >= 7 */ + const FLAC__int32 *const buf[], + FLAC__uint64 nbytes); #endif /* _FLAC_COMMON_H */ diff --git a/src/decoder/_ogg_common.c b/src/decoder/_ogg_common.c index 6c6553422..bd0650ac4 100644 --- a/src/decoder/_ogg_common.c +++ b/src/decoder/_ogg_common.c @@ -1,5 +1,5 @@ /* - * Copyright (C) 2003-2009 The Music Player Daemon Project + * Copyright (C) 2003-2010 The Music Player Daemon Project * http://www.musicpd.org * * This program is free software; you can redistribute it and/or modify @@ -21,8 +21,8 @@ * Common functions used for Ogg data streams (Ogg-Vorbis and OggFLAC) */ +#include "config.h" #include "_ogg_common.h" -#include "../utils.h" ogg_stream_type ogg_stream_type_detect(struct input_stream *inStream) { diff --git a/src/decoder/_ogg_common.h b/src/decoder/_ogg_common.h index e650c366d..f8446c69c 100644 --- a/src/decoder/_ogg_common.h +++ b/src/decoder/_ogg_common.h @@ -1,5 +1,5 @@ /* - * Copyright (C) 2003-2009 The Music Player Daemon Project + * Copyright (C) 2003-2010 The Music Player Daemon Project * http://www.musicpd.org * * This program is free software; you can redistribute it and/or modify @@ -24,7 +24,7 @@ #ifndef MPD_OGG_COMMON_H #define MPD_OGG_COMMON_H -#include "../decoder_api.h" +#include "decoder_api.h" typedef enum _ogg_stream_type { VORBIS, FLAC } ogg_stream_type; diff --git a/src/decoder/audiofile_plugin.c b/src/decoder/audiofile_decoder_plugin.c index f66d90dc1..b099cf706 100644 --- a/src/decoder/audiofile_plugin.c +++ b/src/decoder/audiofile_decoder_plugin.c @@ -1,5 +1,5 @@ /* - * Copyright (C) 2003-2009 The Music Player Daemon Project + * Copyright (C) 2003-2010 The Music Player Daemon Project * http://www.musicpd.org * * This program is free software; you can redistribute it and/or modify @@ -17,7 +17,9 @@ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ -#include "../decoder_api.h" +#include "config.h" +#include "decoder_api.h" +#include "audio_check.h" #include <audiofile.h> #include <af_vfs.h> @@ -45,10 +47,20 @@ static int audiofile_get_duration(const char *file) } static ssize_t -audiofile_file_read(AFvirtualfile *vfile, void *data, size_t nbytes) +audiofile_file_read(AFvirtualfile *vfile, void *data, size_t length) { struct input_stream *is = (struct input_stream *) vfile->closure; - return input_stream_read(is, data, nbytes); + GError *error = NULL; + size_t nbytes; + + nbytes = input_stream_read(is, data, length, &error); + if (nbytes == 0 && error != NULL) { + g_warning("%s", error->message); + g_error_free(error); + return -1; + } + + return nbytes; } static long @@ -78,7 +90,7 @@ audiofile_file_seek(AFvirtualfile *vfile, long offset, int is_relative) { struct input_stream *is = (struct input_stream *) vfile->closure; int whence = (is_relative ? SEEK_CUR : SEEK_SET); - if (input_stream_seek(is, offset, whence)) { + if (input_stream_seek(is, offset, whence, NULL)) { return is->offset; } else { return -1; @@ -99,17 +111,56 @@ setup_virtual_fops(struct input_stream *stream) return vf; } +static enum sample_format +audiofile_bits_to_sample_format(int bits) +{ + switch (bits) { + case 8: + return SAMPLE_FORMAT_S8; + + case 16: + return SAMPLE_FORMAT_S16; + + case 24: + return SAMPLE_FORMAT_S24_P32; + + case 32: + return SAMPLE_FORMAT_S32; + } + + return SAMPLE_FORMAT_UNDEFINED; +} + +static enum sample_format +audiofile_setup_sample_format(AFfilehandle af_fp) +{ + int fs, bits; + + afGetSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits); + if (!audio_valid_sample_format(audiofile_bits_to_sample_format(bits))) { + g_debug("input file has %d bit samples, converting to 16", + bits); + bits = 16; + } + + afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, + AF_SAMPFMT_TWOSCOMP, bits); + afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits); + + return audiofile_bits_to_sample_format(bits); +} + static void audiofile_stream_decode(struct decoder *decoder, struct input_stream *is) { + GError *error = NULL; AFvirtualfile *vf; int fs, frame_count; AFfilehandle af_fp; - int bits; struct audio_format audio_format; float total_time; uint16_t bit_rate; - int ret, current = 0; + int ret; char chunk[CHUNK_SIZE]; enum decoder_command cmd; @@ -126,26 +177,13 @@ audiofile_stream_decode(struct decoder *decoder, struct input_stream *is) return; } - afGetSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits); - if (!audio_valid_sample_format(bits)) { - g_debug("input file has %d bit samples, converting to 16", - bits); - bits = 16; - } - - afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, - AF_SAMPFMT_TWOSCOMP, bits); - afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits); - audio_format.bits = (uint8_t)bits; - audio_format.sample_rate = - (unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK); - audio_format.channels = - (uint8_t)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK); - - if (!audio_format_valid(&audio_format)) { - g_warning("Invalid audio format: %u:%u:%u\n", - audio_format.sample_rate, audio_format.bits, - audio_format.channels); + if (!audio_format_init_checked(&audio_format, + afGetRate(af_fp, AF_DEFAULT_TRACK), + audiofile_setup_sample_format(af_fp), + afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK), + &error)) { + g_warning("%s", error->message); + g_error_free(error); afCloseFile(af_fp); return; } @@ -166,17 +204,14 @@ audiofile_stream_decode(struct decoder *decoder, struct input_stream *is) if (ret <= 0) break; - current += ret; cmd = decoder_data(decoder, NULL, chunk, ret * fs, - (float)current / - (float)audio_format.sample_rate, - bit_rate, NULL); + bit_rate); if (cmd == DECODE_COMMAND_SEEK) { - current = decoder_seek_where(decoder) * + AFframecount frame = decoder_seek_where(decoder) * audio_format.sample_rate; - afSeekFrame(af_fp, AF_DEFAULT_TRACK, current); + afSeekFrame(af_fp, AF_DEFAULT_TRACK, frame); decoder_command_finished(decoder); cmd = DECODE_COMMAND_NONE; @@ -209,7 +244,7 @@ static const char *const audiofile_suffixes[] = { static const char *const audiofile_mime_types[] = { "audio/x-wav", "audio/x-aiff", - NULL + NULL }; const struct decoder_plugin audiofile_decoder_plugin = { diff --git a/src/decoder/faad_plugin.c b/src/decoder/faad_decoder_plugin.c index 7b2806a4c..8f932ad58 100644 --- a/src/decoder/faad_plugin.c +++ b/src/decoder/faad_decoder_plugin.c @@ -1,5 +1,5 @@ /* - * Copyright (C) 2003-2009 The Music Player Daemon Project + * Copyright (C) 2003-2010 The Music Player Daemon Project * http://www.musicpd.org * * This program is free software; you can redistribute it and/or modify @@ -17,9 +17,10 @@ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ -#include "../decoder_api.h" -#include "decoder_buffer.h" #include "config.h" +#include "decoder_api.h" +#include "decoder_buffer.h" +#include "audio_check.h" #define AAC_MAX_CHANNELS 6 @@ -37,6 +38,15 @@ static const unsigned adts_sample_rates[] = }; /** + * The GLib quark used for errors reported by this plugin. + */ +static inline GQuark +faad_decoder_quark(void) +{ + return g_quark_from_static_string("faad"); +} + +/** * Check whether the buffer head is an AAC frame, and return the frame * length. Returns 0 if it is not a frame. */ @@ -195,7 +205,7 @@ faad_song_duration(struct decoder_buffer *buffer, struct input_stream *is) /* obtain the duration from the ADTS header */ float song_length = adts_song_duration(buffer); - input_stream_seek(is, tagsize, SEEK_SET); + input_stream_seek(is, tagsize, SEEK_SET, NULL); data = decoder_buffer_read(buffer, &length); if (data != NULL) @@ -232,7 +242,7 @@ faad_song_duration(struct decoder_buffer *buffer, struct input_stream *is) */ static bool faad_decoder_init(faacDecHandle decoder, struct decoder_buffer *buffer, - struct audio_format *audio_format) + struct audio_format *audio_format, GError **error_r) { union { /* deconst hack for libfaad */ @@ -247,32 +257,33 @@ faad_decoder_init(faacDecHandle decoder, struct decoder_buffer *buffer, /* neaacdec.h declares all arguments as "unsigned long", but internally expects uint32_t pointers. To avoid gcc warnings, use this workaround. */ - unsigned long *sample_rate_r = (unsigned long *)(void *)&sample_rate; + unsigned long *sample_rate_p = (unsigned long *)(void *)&sample_rate; #else - uint32_t *sample_rate_r = &sample_rate; + uint32_t *sample_rate_p = &sample_rate; #endif u.in = decoder_buffer_read(buffer, &length); - if (u.in == NULL) + if (u.in == NULL) { + g_set_error(error_r, faad_decoder_quark(), 0, + "Empty file"); return false; + } nbytes = faacDecInit(decoder, u.out, #ifdef HAVE_FAAD_BUFLEN_FUNCS length, #endif - sample_rate_r, &channels); - if (nbytes < 0) + sample_rate_p, &channels); + if (nbytes < 0) { + g_set_error(error_r, faad_decoder_quark(), 0, + "Not an AAC stream"); return false; + } decoder_buffer_consume(buffer, nbytes); - *audio_format = (struct audio_format){ - .bits = 16, - .channels = channels, - .sample_rate = sample_rate, - }; - - return true; + return audio_format_init_checked(audio_format, sample_rate, + SAMPLE_FORMAT_S16, channels, error_r); } /** @@ -311,20 +322,16 @@ faad_decoder_decode(faacDecHandle decoder, struct decoder_buffer *buffer, * file is invalid. */ static float -faad_get_file_time_float(const char *file) +faad_get_file_time_float(struct input_stream *is) { struct decoder_buffer *buffer; float length; faacDecHandle decoder; faacDecConfigurationPtr config; - struct input_stream is; - - if (!input_stream_open(&is, file)) - return -1; - buffer = decoder_buffer_new(NULL, &is, + buffer = decoder_buffer_new(NULL, is, FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS); - length = faad_song_duration(buffer, &is); + length = faad_song_duration(buffer, is); if (length < 0) { bool ret; @@ -338,15 +345,14 @@ faad_get_file_time_float(const char *file) decoder_buffer_fill(buffer); - ret = faad_decoder_init(decoder, buffer, &audio_format); - if (ret && audio_format_valid(&audio_format)) + ret = faad_decoder_init(decoder, buffer, &audio_format, NULL); + if (ret) length = 0; faacDecClose(decoder); } decoder_buffer_free(buffer); - input_stream_close(&is); return length; } @@ -357,12 +363,12 @@ faad_get_file_time_float(const char *file) * file is invalid. */ static int -faad_get_file_time(const char *file) +faad_get_file_time(struct input_stream *is) { int file_time = -1; float length; - if ((length = faad_get_file_time_float(file)) >= 0) + if ((length = faad_get_file_time_float(is)) >= 0) file_time = length + 0.5; return file_time; @@ -371,7 +377,7 @@ faad_get_file_time(const char *file) static void faad_stream_decode(struct decoder *mpd_decoder, struct input_stream *is) { - float file_time; + GError *error = NULL; float total_time = 0; faacDecHandle decoder; struct audio_format audio_format; @@ -408,15 +414,10 @@ faad_stream_decode(struct decoder *mpd_decoder, struct input_stream *is) /* initialize it */ - ret = faad_decoder_init(decoder, buffer, &audio_format); + ret = faad_decoder_init(decoder, buffer, &audio_format, &error); if (!ret) { - g_warning("Error not a AAC stream.\n"); - faacDecClose(decoder); - return; - } - - if (!audio_format_valid(&audio_format)) { - g_warning("invalid audio format\n"); + g_warning("%s", error->message); + g_error_free(error); faacDecClose(decoder); return; } @@ -427,8 +428,6 @@ faad_stream_decode(struct decoder *mpd_decoder, struct input_stream *is) /* the decoder loop */ - file_time = 0.0; - do { size_t frame_size; const void *decoded; @@ -474,16 +473,13 @@ faad_stream_decode(struct decoder *mpd_decoder, struct input_stream *is) bit_rate = frame_info.bytesconsumed * 8.0 * frame_info.channels * audio_format.sample_rate / frame_info.samples / 1000 + 0.5; - file_time += - (float)(frame_info.samples) / frame_info.channels / - audio_format.sample_rate; } /* send PCM samples to MPD */ cmd = decoder_data(mpd_decoder, is, decoded, - (size_t)frame_info.samples * 2, file_time, - bit_rate, NULL); + (size_t)frame_info.samples * 2, + bit_rate); } while (cmd != DECODE_COMMAND_STOP); /* cleanup */ @@ -492,15 +488,13 @@ faad_stream_decode(struct decoder *mpd_decoder, struct input_stream *is) } static struct tag * -faad_tag_dup(const char *file) +faad_stream_tag(struct input_stream *is) { - int file_time = faad_get_file_time(file); + int file_time = faad_get_file_time(is); struct tag *tag; - if (file_time < 0) { - g_debug("Failed to get total song time from: %s", file); + if (file_time < 0) return NULL; - } tag = tag_new(); tag->time = file_time; @@ -515,7 +509,7 @@ static const char *const faad_mime_types[] = { const struct decoder_plugin faad_decoder_plugin = { .name = "faad", .stream_decode = faad_stream_decode, - .tag_dup = faad_tag_dup, + .stream_tag = faad_stream_tag, .suffixes = faad_suffixes, .mime_types = faad_mime_types, }; diff --git a/src/decoder/ffmpeg_plugin.c b/src/decoder/ffmpeg_decoder_plugin.c index 10894b633..f9d4eb8a9 100644 --- a/src/decoder/ffmpeg_plugin.c +++ b/src/decoder/ffmpeg_decoder_plugin.c @@ -1,5 +1,5 @@ /* - * Copyright (C) 2003-2009 The Music Player Daemon Project + * Copyright (C) 2003-2010 The Music Player Daemon Project * http://www.musicpd.org * * This program is free software; you can redistribute it and/or modify @@ -17,8 +17,9 @@ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ -#include "../decoder_api.h" #include "config.h" +#include "decoder_api.h" +#include "audio_check.h" #include <glib.h> @@ -39,19 +40,46 @@ #include <libavcodec/avcodec.h> #include <libavformat/avformat.h> #include <libavformat/avio.h> +#include <libavutil/log.h> #endif #undef G_LOG_DOMAIN #define G_LOG_DOMAIN "ffmpeg" -struct ffmpeg_context { - int audio_stream; - AVFormatContext *format_context; - AVCodecContext *codec_context; - struct decoder *decoder; - struct input_stream *input; - struct tag *tag; -}; +#ifndef OLD_FFMPEG_INCLUDES + +static GLogLevelFlags +level_ffmpeg_to_glib(int level) +{ + if (level <= AV_LOG_FATAL) + return G_LOG_LEVEL_CRITICAL; + + if (level <= AV_LOG_ERROR) + return G_LOG_LEVEL_WARNING; + + if (level <= AV_LOG_INFO) + return G_LOG_LEVEL_MESSAGE; + + return G_LOG_LEVEL_DEBUG; +} + +static void +mpd_ffmpeg_log_callback(G_GNUC_UNUSED void *ptr, int level, + const char *fmt, va_list vl) +{ + const AVClass * cls = NULL; + + if (ptr != NULL) + cls = *(const AVClass *const*)ptr; + + if (cls != NULL) { + char *domain = g_strconcat(G_LOG_DOMAIN, "/", cls->item_name(ptr), NULL); + g_logv(domain, level_ffmpeg_to_glib(level), fmt, vl); + g_free(domain); + } +} + +#endif /* !OLD_FFMPEG_INCLUDES */ struct mpd_ffmpeg_stream { struct decoder *decoder; @@ -79,7 +107,7 @@ mpd_ffmpeg_stream_seek(void *opaque, int64_t pos, int whence) if (whence == AVSEEK_SIZE) return stream->input->size; - ret = input_stream_seek(stream->input, pos, whence); + ret = input_stream_seek(stream->input, pos, whence, NULL); if (!ret) return -1; @@ -115,6 +143,10 @@ mpd_ffmpeg_stream_close(struct mpd_ffmpeg_stream *stream) static bool ffmpeg_init(G_GNUC_UNUSED const struct config_param *param) { +#ifndef OLD_FFMPEG_INCLUDES + av_log_set_callback(mpd_ffmpeg_log_callback); +#endif + av_register_all(); return true; } @@ -130,9 +162,95 @@ ffmpeg_find_audio_stream(const AVFormatContext *format_context) return -1; } +/** + * On some platforms, libavcodec wants the output buffer aligned to 16 + * bytes (because it uses SSE/Altivec internally). This function + * returns the aligned version of the specified buffer, and corrects + * the buffer size. + */ +static void * +align16(void *p, size_t *length_p) +{ + unsigned add = 16 - (size_t)p % 16; + + *length_p -= add; + return (char *)p + add; +} + +static enum decoder_command +ffmpeg_send_packet(struct decoder *decoder, struct input_stream *is, + const AVPacket *packet, + AVCodecContext *codec_context, + const AVRational *time_base) +{ + enum decoder_command cmd = DECODE_COMMAND_NONE; + uint8_t audio_buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2 + 16]; + int16_t *aligned_buffer; + size_t buffer_size; + int len, audio_size; + uint8_t *packet_data; + int packet_size; + + if (packet->pts != (int64_t)AV_NOPTS_VALUE) + decoder_timestamp(decoder, + av_rescale_q(packet->pts, *time_base, + (AVRational){1, 1})); + + packet_data = packet->data; + packet_size = packet->size; + + buffer_size = sizeof(audio_buf); + aligned_buffer = align16(audio_buf, &buffer_size); + + while ((packet_size > 0) && (cmd == DECODE_COMMAND_NONE)) { + audio_size = buffer_size; + len = avcodec_decode_audio2(codec_context, + aligned_buffer, &audio_size, + packet_data, packet_size); + + if (len < 0) { + /* if error, we skip the frame */ + g_message("decoding failed\n"); + break; + } + + packet_data += len; + packet_size -= len; + + if (audio_size <= 0) + continue; + + cmd = decoder_data(decoder, is, + aligned_buffer, audio_size, + codec_context->bit_rate / 1000); + } + return cmd; +} + +static enum sample_format +ffmpeg_sample_format(G_GNUC_UNUSED const AVCodecContext *codec_context) +{ +#if LIBAVCODEC_VERSION_INT >= ((51<<16)+(41<<8)+0) + switch (codec_context->sample_fmt) { + case SAMPLE_FMT_S16: + return SAMPLE_FORMAT_S16; + + case SAMPLE_FMT_S32: + return SAMPLE_FORMAT_S32; + + default: + g_warning("Unsupported libavcodec SampleFormat value: %d", + codec_context->sample_fmt); + return SAMPLE_FORMAT_UNDEFINED; + } +#else + /* XXX fixme 16-bit for older ffmpeg (13 Aug 2007) */ + return SAMPLE_FORMAT_S16; +#endif +} + static AVInputFormat * -ffmpeg_probe(struct decoder *decoder, struct input_stream *is, - const char *uri) +ffmpeg_probe(struct decoder *decoder, struct input_stream *is) { enum { BUFFER_SIZE = 16384, @@ -141,7 +259,7 @@ ffmpeg_probe(struct decoder *decoder, struct input_stream *is, unsigned char *buffer = g_malloc(BUFFER_SIZE); size_t nbytes = decoder_read(decoder, is, buffer, BUFFER_SIZE); - if (nbytes <= PADDING || !input_stream_seek(is, 0, SEEK_SET)) { + if (nbytes <= PADDING || !input_stream_seek(is, 0, SEEK_SET, NULL)) { g_free(buffer); return NULL; } @@ -155,7 +273,7 @@ ffmpeg_probe(struct decoder *decoder, struct input_stream *is, AVProbeData avpd = { .buf = buffer, .buf_size = nbytes, - .filename = uri, + .filename = is->uri, }; AVInputFormat *format = av_probe_input_format(&avpd, true); @@ -164,15 +282,12 @@ ffmpeg_probe(struct decoder *decoder, struct input_stream *is, return format; } -static bool -ffmpeg_helper(const char *uri, - struct decoder *decoder, struct input_stream *input, - bool (*callback)(struct ffmpeg_context *ctx), - struct ffmpeg_context *ctx) +static void +ffmpeg_decode(struct decoder *decoder, struct input_stream *input) { - AVInputFormat *input_format = ffmpeg_probe(decoder, input, uri); + AVInputFormat *input_format = ffmpeg_probe(decoder, input); if (input_format == NULL) - return false; + return; g_debug("detected input format '%s' (%s)", input_format->name, input_format->long_name); @@ -181,28 +296,27 @@ ffmpeg_helper(const char *uri, mpd_ffmpeg_stream_open(decoder, input); if (stream == NULL) { g_warning("Failed to open stream"); - return false; + return; } AVFormatContext *format_context; AVCodecContext *codec_context; AVCodec *codec; int audio_stream; - bool ret; //ffmpeg works with ours "fileops" helper - if (av_open_input_stream(&format_context, stream->io, uri, + if (av_open_input_stream(&format_context, stream->io, input->uri, input_format, NULL) != 0) { g_warning("Open failed\n"); mpd_ffmpeg_stream_close(stream); - return false; + return; } if (av_find_stream_info(format_context)<0) { g_warning("Couldn't find stream info\n"); av_close_input_stream(format_context); mpd_ffmpeg_stream_close(stream); - return false; + return; } audio_stream = ffmpeg_find_audio_stream(format_context); @@ -210,7 +324,7 @@ ffmpeg_helper(const char *uri, g_warning("No audio stream inside\n"); av_close_input_stream(format_context); mpd_ffmpeg_stream_close(stream); - return false; + return; } codec_context = format_context->streams[audio_stream]->codec; @@ -223,145 +337,48 @@ ffmpeg_helper(const char *uri, g_warning("Unsupported audio codec\n"); av_close_input_stream(format_context); mpd_ffmpeg_stream_close(stream); - return false; + return; } if (avcodec_open(codec_context, codec)<0) { g_warning("Could not open codec\n"); av_close_input_stream(format_context); mpd_ffmpeg_stream_close(stream); - return false; - } - - if (callback) { - ctx->audio_stream = audio_stream; - ctx->format_context = format_context; - ctx->codec_context = codec_context; - - ret = callback(ctx); - } else - ret = true; - - avcodec_close(codec_context); - av_close_input_stream(format_context); - mpd_ffmpeg_stream_close(stream); - - return ret; -} - -/** - * On some platforms, libavcodec wants the output buffer aligned to 16 - * bytes (because it uses SSE/Altivec internally). This function - * returns the aligned version of the specified buffer, and corrects - * the buffer size. - */ -static void * -align16(void *p, size_t *length_p) -{ - unsigned add = 16 - (size_t)p % 16; - - *length_p -= add; - return (char *)p + add; -} - -static enum decoder_command -ffmpeg_send_packet(struct decoder *decoder, struct input_stream *is, - const AVPacket *packet, - AVCodecContext *codec_context, - const AVRational *time_base) -{ - enum decoder_command cmd = DECODE_COMMAND_NONE; - int position; - uint8_t audio_buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2 + 16]; - int16_t *aligned_buffer; - size_t buffer_size; - int len, audio_size; - uint8_t *packet_data; - int packet_size; - - packet_data = packet->data; - packet_size = packet->size; - - buffer_size = sizeof(audio_buf); - aligned_buffer = align16(audio_buf, &buffer_size); - - while ((packet_size > 0) && (cmd == DECODE_COMMAND_NONE)) { - audio_size = buffer_size; - len = avcodec_decode_audio2(codec_context, - aligned_buffer, &audio_size, - packet_data, packet_size); - - if (len < 0) { - /* if error, we skip the frame */ - g_message("decoding failed\n"); - break; - } - - packet_data += len; - packet_size -= len; - - if (audio_size <= 0) - continue; - - position = packet->pts != (int64_t)AV_NOPTS_VALUE - ? av_rescale_q(packet->pts, *time_base, - (AVRational){1, 1}) - : 0; - - cmd = decoder_data(decoder, is, - aligned_buffer, audio_size, - position, - codec_context->bit_rate / 1000, NULL); + return; } - return cmd; -} -static bool -ffmpeg_decode_internal(struct ffmpeg_context *ctx) -{ - struct decoder *decoder = ctx->decoder; - AVCodecContext *codec_context = ctx->codec_context; - AVFormatContext *format_context = ctx->format_context; - AVPacket packet; + GError *error = NULL; struct audio_format audio_format; - enum decoder_command cmd; - int total_time; - - total_time = 0; - -#if LIBAVCODEC_VERSION_INT >= ((51<<16)+(41<<8)+0) - audio_format.bits = (uint8_t) av_get_bits_per_sample_format(codec_context->sample_fmt); -#else - /* XXX fixme 16-bit for older ffmpeg (13 Aug 2007) */ - audio_format.bits = (uint8_t) 16; -#endif - audio_format.sample_rate = (unsigned int)codec_context->sample_rate; - audio_format.channels = codec_context->channels; - - if (!audio_format_valid(&audio_format)) { - g_warning("Invalid audio format: %u:%u:%u\n", - audio_format.sample_rate, audio_format.bits, - audio_format.channels); - return false; + if (!audio_format_init_checked(&audio_format, + codec_context->sample_rate, + ffmpeg_sample_format(codec_context), + codec_context->channels, &error)) { + g_warning("%s", error->message); + g_error_free(error); + avcodec_close(codec_context); + av_close_input_stream(format_context); + mpd_ffmpeg_stream_close(stream); + return; } - //there is some problem with this on some demux (mp3 at least) - if (format_context->duration != (int64_t)AV_NOPTS_VALUE) { - total_time = format_context->duration / AV_TIME_BASE; - } + int total_time = format_context->duration != (int64_t)AV_NOPTS_VALUE + ? format_context->duration / AV_TIME_BASE + : 0; decoder_initialized(decoder, &audio_format, - ctx->input->seekable, total_time); + input->seekable, total_time); + enum decoder_command cmd; do { + AVPacket packet; if (av_read_frame(format_context, &packet) < 0) /* end of file */ break; - if (packet.stream_index == ctx->audio_stream) - cmd = ffmpeg_send_packet(decoder, ctx->input, + if (packet.stream_index == audio_stream) + cmd = ffmpeg_send_packet(decoder, input, &packet, codec_context, - &format_context->streams[ctx->audio_stream]->time_base); + &format_context->streams[audio_stream]->time_base); else cmd = decoder_get_command(decoder); @@ -378,115 +395,121 @@ ffmpeg_decode_internal(struct ffmpeg_context *ctx) } } while (cmd != DECODE_COMMAND_STOP); - return true; + avcodec_close(codec_context); + av_close_input_stream(format_context); + mpd_ffmpeg_stream_close(stream); } -static void -ffmpeg_decode(struct decoder *decoder, struct input_stream *input) -{ - struct ffmpeg_context ctx; - - ctx.input = input; - ctx.decoder = decoder; +#if LIBAVFORMAT_VERSION_INT >= ((52<<16)+(31<<8)+0) +typedef struct ffmpeg_tag_map { + enum tag_type type; + const char *name; +} ffmpeg_tag_map; - char *uri = decoder_get_uri(decoder); - ffmpeg_helper(uri, decoder, input, - ffmpeg_decode_internal, &ctx); - g_free(uri); -} +static const ffmpeg_tag_map ffmpeg_tag_maps[] = { + { TAG_TITLE, "title" }, +#if LIBAVFORMAT_VERSION_INT >= ((52<<16)+(50<<8)) + { TAG_ARTIST, "artist" }, + { TAG_DATE, "date" }, +#else + { TAG_ARTIST, "author" }, + { TAG_DATE, "year" }, +#endif + { TAG_ALBUM, "album" }, + { TAG_COMMENT, "comment" }, + { TAG_GENRE, "genre" }, + { TAG_TRACK, "track" }, + { TAG_ARTIST_SORT, "author-sort" }, + { TAG_ALBUM_ARTIST, "album_artist" }, + { TAG_ALBUM_ARTIST_SORT, "album_artist-sort" }, + { TAG_COMPOSER, "composer" }, + { TAG_PERFORMER, "performer" }, + { TAG_DISC, "disc" }, +}; -#if LIBAVFORMAT_VERSION_INT >= ((52<<16)+(31<<8)+0) static bool ffmpeg_copy_metadata(struct tag *tag, AVMetadata *m, - enum tag_type type, const char *name) + const ffmpeg_tag_map tag_map) { - AVMetadataTag *mt = av_metadata_get(m, name, NULL, 0); - if (mt != NULL) - tag_add_item(tag, type, mt->value); + AVMetadataTag *mt = NULL; + + while ((mt = av_metadata_get(m, tag_map.name, mt, 0)) != NULL) + tag_add_item(tag, tag_map.type, mt->value); return mt != NULL; } + #endif -static bool ffmpeg_tag_internal(struct ffmpeg_context *ctx) +//no tag reading in ffmpeg, check if playable +static struct tag * +ffmpeg_stream_tag(struct input_stream *is) { - struct tag *tag = (struct tag *) ctx->tag; - AVFormatContext *f = ctx->format_context; + AVInputFormat *input_format = ffmpeg_probe(NULL, is); + if (input_format == NULL) + return NULL; - tag->time = 0; - if (f->duration != (int64_t)AV_NOPTS_VALUE) - tag->time = f->duration / AV_TIME_BASE; + struct mpd_ffmpeg_stream *stream = mpd_ffmpeg_stream_open(NULL, is); + if (stream == NULL) + return NULL; + + AVFormatContext *f; + if (av_open_input_stream(&f, stream->io, is->uri, + input_format, NULL) != 0) { + mpd_ffmpeg_stream_close(stream); + return NULL; + } + + if (av_find_stream_info(f) < 0) { + av_close_input_stream(f); + mpd_ffmpeg_stream_close(stream); + return NULL; + } + + struct tag *tag = tag_new(); + + tag->time = f->duration != (int64_t)AV_NOPTS_VALUE + ? f->duration / AV_TIME_BASE + : 0; #if LIBAVFORMAT_VERSION_INT >= ((52<<16)+(31<<8)+0) av_metadata_conv(f, NULL, f->iformat->metadata_conv); - ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_TITLE, "title"); -#if LIBAVFORMAT_VERSION_INT >= ((52<<16)+(50<<8)) - ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_ARTIST, "artist"); - ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_DATE, "date"); -#else - ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_ARTIST, "author"); - ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_DATE, "year"); -#endif - ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_ALBUM, "album"); - ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_COMMENT, "comment"); - ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_GENRE, "genre"); - ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_TRACK, "track"); - ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_ALBUM_ARTIST, "album_artist"); - ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_COMPOSER, "composer"); - ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_PERFORMER, "performer"); - ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_DISC, "disc"); + for (unsigned i = 0; i < sizeof(ffmpeg_tag_maps)/sizeof(ffmpeg_tag_map); i++) { + int idx = ffmpeg_find_audio_stream(f); + ffmpeg_copy_metadata(tag, f->metadata, ffmpeg_tag_maps[i]); + if (idx >= 0) + ffmpeg_copy_metadata(tag, f->streams[idx]->metadata, ffmpeg_tag_maps[i]); + } #else if (f->author[0]) - tag_add_item(tag, TAG_ITEM_ARTIST, f->author); + tag_add_item(tag, TAG_ARTIST, f->author); if (f->title[0]) - tag_add_item(tag, TAG_ITEM_TITLE, f->title); + tag_add_item(tag, TAG_TITLE, f->title); if (f->album[0]) - tag_add_item(tag, TAG_ITEM_ALBUM, f->album); + tag_add_item(tag, TAG_ALBUM, f->album); if (f->track > 0) { char buffer[16]; snprintf(buffer, sizeof(buffer), "%d", f->track); - tag_add_item(tag, TAG_ITEM_TRACK, buffer); + tag_add_item(tag, TAG_TRACK, buffer); } if (f->comment[0]) - tag_add_item(tag, TAG_ITEM_COMMENT, f->comment); + tag_add_item(tag, TAG_COMMENT, f->comment); if (f->genre[0]) - tag_add_item(tag, TAG_ITEM_GENRE, f->genre); + tag_add_item(tag, TAG_GENRE, f->genre); if (f->year > 0) { char buffer[16]; snprintf(buffer, sizeof(buffer), "%d", f->year); - tag_add_item(tag, TAG_ITEM_DATE, buffer); + tag_add_item(tag, TAG_DATE, buffer); } #endif - return true; -} - -//no tag reading in ffmpeg, check if playable -static struct tag *ffmpeg_tag(const char *file) -{ - struct input_stream input; - struct ffmpeg_context ctx; - bool ret; - - if (!input_stream_open(&input, file)) { - g_warning("failed to open %s\n", file); - return NULL; - } - - ctx.decoder = NULL; - ctx.tag = tag_new(); - - ret = ffmpeg_helper(file, NULL, &input, ffmpeg_tag_internal, &ctx); - if (!ret) { - tag_free(ctx.tag); - ctx.tag = NULL; - } - input_stream_close(&input); + av_close_input_stream(f); + mpd_ffmpeg_stream_close(stream); - return ctx.tag; + return tag; } /** @@ -592,6 +615,12 @@ static const char *const ffmpeg_mime_types[] = { "video/x-vid", "video/x-wmv", "video/x-xvid", + + /* special value for the "ffmpeg" input plugin: all streams by + the "ffmpeg" input plugin shall be decoded by this + plugin */ + "audio/x-mpd-ffmpeg", + NULL }; @@ -599,7 +628,7 @@ const struct decoder_plugin ffmpeg_decoder_plugin = { .name = "ffmpeg", .init = ffmpeg_init, .stream_decode = ffmpeg_decode, - .tag_dup = ffmpeg_tag, + .stream_tag = ffmpeg_stream_tag, .suffixes = ffmpeg_suffixes, .mime_types = ffmpeg_mime_types }; diff --git a/src/decoder/flac_compat.h b/src/decoder/flac_compat.h new file mode 100644 index 000000000..d597690a0 --- /dev/null +++ b/src/decoder/flac_compat.h @@ -0,0 +1,114 @@ +/* + * Copyright (C) 2003-2010 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. + */ + +/* + * Common data structures and functions used by FLAC and OggFLAC + */ + +#ifndef MPD_FLAC_COMPAT_H +#define MPD_FLAC_COMPAT_H + +#include <FLAC/export.h> +#if !defined(FLAC_API_VERSION_CURRENT) || FLAC_API_VERSION_CURRENT <= 7 +# include <FLAC/seekable_stream_decoder.h> + +/* starting with libFLAC 1.1.3, the SeekableStreamDecoder has been + merged into the StreamDecoder. The following macros try to emulate + the new API for libFLAC 1.1.2 by mapping MPD's StreamDecoder calls + to the old SeekableStreamDecoder API. */ + +#define FLAC__StreamDecoder FLAC__SeekableStreamDecoder +#define FLAC__stream_decoder_new FLAC__seekable_stream_decoder_new +#define FLAC__stream_decoder_get_decode_position FLAC__seekable_stream_decoder_get_decode_position +#define FLAC__stream_decoder_get_state FLAC__seekable_stream_decoder_get_state +#define FLAC__stream_decoder_process_single FLAC__seekable_stream_decoder_process_single +#define FLAC__stream_decoder_process_until_end_of_metadata FLAC__seekable_stream_decoder_process_until_end_of_metadata +#define FLAC__stream_decoder_seek_absolute FLAC__seekable_stream_decoder_seek_absolute +#define FLAC__stream_decoder_finish FLAC__seekable_stream_decoder_finish +#define FLAC__stream_decoder_delete FLAC__seekable_stream_decoder_delete + +#define FLAC__STREAM_DECODER_END_OF_STREAM FLAC__SEEKABLE_STREAM_DECODER_END_OF_STREAM + +typedef unsigned flac_read_status_size_t; + +#define FLAC__StreamDecoderReadStatus FLAC__SeekableStreamDecoderReadStatus +#define FLAC__STREAM_DECODER_READ_STATUS_CONTINUE FLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_OK +#define FLAC__STREAM_DECODER_READ_STATUS_END_OF_STREAM FLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_OK +#define FLAC__STREAM_DECODER_READ_STATUS_ABORT FLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_ERROR + +#define FLAC__StreamDecoderSeekStatus FLAC__SeekableStreamDecoderSeekStatus +#define FLAC__STREAM_DECODER_SEEK_STATUS_OK FLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_OK +#define FLAC__STREAM_DECODER_SEEK_STATUS_ERROR FLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_ERROR +#define FLAC__STREAM_DECODER_SEEK_STATUS_UNSUPPORTED FLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_ERROR + +#define FLAC__StreamDecoderTellStatus FLAC__SeekableStreamDecoderTellStatus +#define FLAC__STREAM_DECODER_TELL_STATUS_OK FLAC__SEEKABLE_STREAM_DECODER_TELL_STATUS_OK +#define FLAC__STREAM_DECODER_TELL_STATUS_ERROR FLAC__SEEKABLE_STREAM_DECODER_TELL_STATUS_ERROR +#define FLAC__STREAM_DECODER_TELL_STATUS_UNSUPPORTED FLAC__SEEKABLE_STREAM_DECODER_TELL_STATUS_ERROR + +#define FLAC__StreamDecoderLengthStatus FLAC__SeekableStreamDecoderLengthStatus +#define FLAC__STREAM_DECODER_LENGTH_STATUS_OK FLAC__SEEKABLE_STREAM_DECODER_LENGTH_STATUS_OK +#define FLAC__STREAM_DECODER_LENGTH_STATUS_ERROR FLAC__SEEKABLE_STREAM_DECODER_LENGTH_STATUS_ERROR +#define FLAC__STREAM_DECODER_LENGTH_STATUS_UNSUPPORTED FLAC__SEEKABLE_STREAM_DECODER_LENGTH_STATUS_ERROR + +typedef enum { + FLAC__STREAM_DECODER_INIT_STATUS_OK, + FLAC__STREAM_DECODER_INIT_STATUS_ERROR, +} FLAC__StreamDecoderInitStatus; + +static inline FLAC__StreamDecoderInitStatus +FLAC__stream_decoder_init_stream(FLAC__SeekableStreamDecoder *decoder, + FLAC__SeekableStreamDecoderReadCallback read_cb, + FLAC__SeekableStreamDecoderSeekCallback seek_cb, + FLAC__SeekableStreamDecoderTellCallback tell_cb, + FLAC__SeekableStreamDecoderLengthCallback length_cb, + FLAC__SeekableStreamDecoderEofCallback eof_cb, + FLAC__SeekableStreamDecoderWriteCallback write_cb, + FLAC__SeekableStreamDecoderMetadataCallback metadata_cb, + FLAC__SeekableStreamDecoderErrorCallback error_cb, + void *data) +{ + return FLAC__seekable_stream_decoder_set_read_callback(decoder, read_cb) && + FLAC__seekable_stream_decoder_set_seek_callback(decoder, seek_cb) && + FLAC__seekable_stream_decoder_set_tell_callback(decoder, tell_cb) && + FLAC__seekable_stream_decoder_set_length_callback(decoder, length_cb) && + FLAC__seekable_stream_decoder_set_eof_callback(decoder, eof_cb) && + FLAC__seekable_stream_decoder_set_write_callback(decoder, write_cb) && + FLAC__seekable_stream_decoder_set_metadata_callback(decoder, metadata_cb) && + FLAC__seekable_stream_decoder_set_metadata_respond(decoder, FLAC__METADATA_TYPE_VORBIS_COMMENT) && + FLAC__seekable_stream_decoder_set_error_callback(decoder, error_cb) && + FLAC__seekable_stream_decoder_set_client_data(decoder, data) && + FLAC__seekable_stream_decoder_init(decoder) == FLAC__SEEKABLE_STREAM_DECODER_OK + ? FLAC__STREAM_DECODER_INIT_STATUS_OK + : FLAC__STREAM_DECODER_INIT_STATUS_ERROR; +} + +#else /* FLAC_API_VERSION_CURRENT > 7 */ + +# include <FLAC/stream_decoder.h> + +# define flac_init(a,b,c,d,e,f,g,h,i,j) \ + (FLAC__stream_decoder_init_stream(a,b,c,d,e,f,g,h,i,j) \ + == FLAC__STREAM_DECODER_INIT_STATUS_OK) + +typedef size_t flac_read_status_size_t; + +#endif /* FLAC_API_VERSION_CURRENT >= 7 */ + +#endif /* _FLAC_COMMON_H */ diff --git a/src/decoder/flac_decoder_plugin.c b/src/decoder/flac_decoder_plugin.c new file mode 100644 index 000000000..9d980b79d --- /dev/null +++ b/src/decoder/flac_decoder_plugin.c @@ -0,0 +1,497 @@ +/* + * Copyright (C) 2003-2010 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" /* must be first for large file support */ +#include "_flac_common.h" +#include "flac_compat.h" +#include "flac_metadata.h" + +#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7 +#include "_ogg_common.h" +#endif + +#include <glib.h> + +#include <assert.h> +#include <unistd.h> + +#include <sys/stat.h> +#include <sys/types.h> + +/* this code was based on flac123, from flac-tools */ + +static FLAC__StreamDecoderReadStatus +flac_read_cb(G_GNUC_UNUSED const FLAC__StreamDecoder *fd, + FLAC__byte buf[], flac_read_status_size_t *bytes, + void *fdata) +{ + struct flac_data *data = fdata; + size_t r; + + r = decoder_read(data->decoder, data->input_stream, + (void *)buf, *bytes); + *bytes = r; + + if (r == 0) { + if (decoder_get_command(data->decoder) != DECODE_COMMAND_NONE || + input_stream_eof(data->input_stream)) + return FLAC__STREAM_DECODER_READ_STATUS_END_OF_STREAM; + else + return FLAC__STREAM_DECODER_READ_STATUS_ABORT; + } + + return FLAC__STREAM_DECODER_READ_STATUS_CONTINUE; +} + +static FLAC__StreamDecoderSeekStatus +flac_seek_cb(G_GNUC_UNUSED const FLAC__StreamDecoder *fd, + FLAC__uint64 offset, void *fdata) +{ + struct flac_data *data = (struct flac_data *) fdata; + + if (!data->input_stream->seekable) + return FLAC__STREAM_DECODER_SEEK_STATUS_UNSUPPORTED; + + if (!input_stream_seek(data->input_stream, offset, SEEK_SET, NULL)) + return FLAC__STREAM_DECODER_SEEK_STATUS_ERROR; + + return FLAC__STREAM_DECODER_SEEK_STATUS_OK; +} + +static FLAC__StreamDecoderTellStatus +flac_tell_cb(G_GNUC_UNUSED const FLAC__StreamDecoder *fd, + FLAC__uint64 * offset, void *fdata) +{ + struct flac_data *data = (struct flac_data *) fdata; + + if (!data->input_stream->seekable) + return FLAC__STREAM_DECODER_TELL_STATUS_UNSUPPORTED; + + *offset = (long)(data->input_stream->offset); + + return FLAC__STREAM_DECODER_TELL_STATUS_OK; +} + +static FLAC__StreamDecoderLengthStatus +flac_length_cb(G_GNUC_UNUSED const FLAC__StreamDecoder *fd, + FLAC__uint64 * length, void *fdata) +{ + struct flac_data *data = (struct flac_data *) fdata; + + if (data->input_stream->size < 0) + return FLAC__STREAM_DECODER_LENGTH_STATUS_UNSUPPORTED; + + *length = (size_t) (data->input_stream->size); + + return FLAC__STREAM_DECODER_LENGTH_STATUS_OK; +} + +static FLAC__bool +flac_eof_cb(G_GNUC_UNUSED const FLAC__StreamDecoder *fd, void *fdata) +{ + struct flac_data *data = (struct flac_data *) fdata; + + return (decoder_get_command(data->decoder) != DECODE_COMMAND_NONE && + decoder_get_command(data->decoder) != DECODE_COMMAND_SEEK) || + input_stream_eof(data->input_stream); +} + +static void +flac_error_cb(G_GNUC_UNUSED const FLAC__StreamDecoder *fd, + FLAC__StreamDecoderErrorStatus status, void *fdata) +{ + flac_error_common_cb("flac", status, (struct flac_data *) fdata); +} + +#if !defined(FLAC_API_VERSION_CURRENT) || FLAC_API_VERSION_CURRENT <= 7 +static void flacPrintErroredState(FLAC__SeekableStreamDecoderState state) +{ + const char *str = ""; /* "" to silence compiler warning */ + switch (state) { + case FLAC__SEEKABLE_STREAM_DECODER_OK: + case FLAC__SEEKABLE_STREAM_DECODER_SEEKING: + case FLAC__SEEKABLE_STREAM_DECODER_END_OF_STREAM: + return; + case FLAC__SEEKABLE_STREAM_DECODER_MEMORY_ALLOCATION_ERROR: + str = "allocation error"; + break; + case FLAC__SEEKABLE_STREAM_DECODER_READ_ERROR: + str = "read error"; + break; + case FLAC__SEEKABLE_STREAM_DECODER_SEEK_ERROR: + str = "seek error"; + break; + case FLAC__SEEKABLE_STREAM_DECODER_STREAM_DECODER_ERROR: + str = "seekable stream error"; + break; + case FLAC__SEEKABLE_STREAM_DECODER_ALREADY_INITIALIZED: + str = "decoder already initialized"; + break; + case FLAC__SEEKABLE_STREAM_DECODER_INVALID_CALLBACK: + str = "invalid callback"; + break; + case FLAC__SEEKABLE_STREAM_DECODER_UNINITIALIZED: + str = "decoder uninitialized"; + } + + g_warning("%s\n", str); +} +#else /* FLAC_API_VERSION_CURRENT >= 7 */ +static void flacPrintErroredState(FLAC__StreamDecoderState state) +{ + const char *str = ""; /* "" to silence compiler warning */ + switch (state) { + case FLAC__STREAM_DECODER_SEARCH_FOR_METADATA: + case FLAC__STREAM_DECODER_READ_METADATA: + case FLAC__STREAM_DECODER_SEARCH_FOR_FRAME_SYNC: + case FLAC__STREAM_DECODER_READ_FRAME: + case FLAC__STREAM_DECODER_END_OF_STREAM: + return; + case FLAC__STREAM_DECODER_OGG_ERROR: + str = "error in the Ogg layer"; + break; + case FLAC__STREAM_DECODER_SEEK_ERROR: + str = "seek error"; + break; + case FLAC__STREAM_DECODER_ABORTED: + str = "decoder aborted by read"; + break; + case FLAC__STREAM_DECODER_MEMORY_ALLOCATION_ERROR: + str = "allocation error"; + break; + case FLAC__STREAM_DECODER_UNINITIALIZED: + str = "decoder uninitialized"; + } + + g_warning("%s\n", str); +} +#endif /* FLAC_API_VERSION_CURRENT >= 7 */ + +static void flacMetadata(G_GNUC_UNUSED const FLAC__StreamDecoder * dec, + const FLAC__StreamMetadata * block, void *vdata) +{ + flac_metadata_common_cb(block, (struct flac_data *) vdata); +} + +static FLAC__StreamDecoderWriteStatus +flac_write_cb(const FLAC__StreamDecoder *dec, const FLAC__Frame *frame, + const FLAC__int32 *const buf[], void *vdata) +{ + struct flac_data *data = (struct flac_data *) vdata; + FLAC__uint64 nbytes = 0; + + if (FLAC__stream_decoder_get_decode_position(dec, &nbytes)) { + if (data->position > 0 && nbytes > data->position) { + nbytes -= data->position; + data->position += nbytes; + } else { + data->position = nbytes; + nbytes = 0; + } + } else + nbytes = 0; + + return flac_common_write(data, frame, buf, nbytes); +} + +static struct tag * +flac_tag_dup(const char *file) +{ + return flac_tag_load(file, NULL); +} + +/** + * Some glue code around FLAC__stream_decoder_new(). + */ +static FLAC__StreamDecoder * +flac_decoder_new(void) +{ + FLAC__StreamDecoder *sd = FLAC__stream_decoder_new(); + if (sd == NULL) { + g_warning("FLAC__stream_decoder_new() failed"); + return NULL; + } + +#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7 + if(!FLAC__stream_decoder_set_metadata_respond(sd, FLAC__METADATA_TYPE_VORBIS_COMMENT)) + g_debug("FLAC__stream_decoder_set_metadata_respond() has failed"); +#endif + + return sd; +} + +static bool +flac_decoder_initialize(struct flac_data *data, FLAC__StreamDecoder *sd, + FLAC__uint64 duration) +{ + data->total_frames = duration; + + if (!FLAC__stream_decoder_process_until_end_of_metadata(sd)) { + g_warning("problem reading metadata"); + return false; + } + + if (data->initialized) { + /* done */ + decoder_initialized(data->decoder, &data->audio_format, + data->input_stream->seekable, + (float)data->total_frames / + (float)data->audio_format.sample_rate); + return true; + } + + if (data->input_stream->seekable) + /* allow the workaround below only for nonseekable + streams*/ + return false; + + /* no stream_info packet found; try to initialize the decoder + from the first frame header */ + FLAC__stream_decoder_process_single(sd); + return data->initialized; +} + +static void +flac_decoder_loop(struct flac_data *data, FLAC__StreamDecoder *flac_dec, + FLAC__uint64 t_start, FLAC__uint64 t_end) +{ + struct decoder *decoder = data->decoder; + enum decoder_command cmd; + + data->first_frame = t_start; + + while (true) { + if (data->tag != NULL && !tag_is_empty(data->tag)) { + cmd = decoder_tag(data->decoder, data->input_stream, + data->tag); + tag_free(data->tag); + data->tag = tag_new(); + } else + cmd = decoder_get_command(decoder); + + if (cmd == DECODE_COMMAND_SEEK) { + FLAC__uint64 seek_sample = t_start + + decoder_seek_where(decoder) * + data->audio_format.sample_rate; + if (seek_sample >= t_start && + (t_end == 0 || seek_sample <= t_end) && + FLAC__stream_decoder_seek_absolute(flac_dec, seek_sample)) { + data->next_frame = seek_sample; + data->position = 0; + decoder_command_finished(decoder); + } else + decoder_seek_error(decoder); + } else if (cmd == DECODE_COMMAND_STOP || + FLAC__stream_decoder_get_state(flac_dec) == FLAC__STREAM_DECODER_END_OF_STREAM) + break; + + if (t_end != 0 && data->next_frame >= t_end) + /* end of this sub track */ + break; + + if (!FLAC__stream_decoder_process_single(flac_dec)) { + cmd = decoder_get_command(decoder); + if (cmd != DECODE_COMMAND_SEEK) + break; + } + } + + if (cmd != DECODE_COMMAND_STOP) { + flacPrintErroredState(FLAC__stream_decoder_get_state(flac_dec)); + FLAC__stream_decoder_finish(flac_dec); + } +} + +static void +flac_decode_internal(struct decoder * decoder, + struct input_stream *input_stream, + bool is_ogg) +{ + FLAC__StreamDecoder *flac_dec; + struct flac_data data; + const char *err = NULL; + + flac_dec = flac_decoder_new(); + if (flac_dec == NULL) + return; + + flac_data_init(&data, decoder, input_stream); + data.tag = tag_new(); + + if (is_ogg) { +#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7 + FLAC__StreamDecoderInitStatus status = + FLAC__stream_decoder_init_ogg_stream(flac_dec, + flac_read_cb, + flac_seek_cb, + flac_tell_cb, + flac_length_cb, + flac_eof_cb, + flac_write_cb, + flacMetadata, + flac_error_cb, + (void *)&data); + if (status != FLAC__STREAM_DECODER_INIT_STATUS_OK) { + err = "doing Ogg init()"; + goto fail; + } +#else + goto fail; +#endif + } else { + FLAC__StreamDecoderInitStatus status = + FLAC__stream_decoder_init_stream(flac_dec, + flac_read_cb, + flac_seek_cb, + flac_tell_cb, + flac_length_cb, + flac_eof_cb, + flac_write_cb, + flacMetadata, + flac_error_cb, + (void *)&data); + if (status != FLAC__STREAM_DECODER_INIT_STATUS_OK) { + err = "doing init()"; + goto fail; + } + } + + if (!flac_decoder_initialize(&data, flac_dec, 0)) { + flac_data_deinit(&data); + FLAC__stream_decoder_delete(flac_dec); + return; + } + + flac_decoder_loop(&data, flac_dec, 0, 0); + +fail: + flac_data_deinit(&data); + FLAC__stream_decoder_delete(flac_dec); + + if (err) + g_warning("%s\n", err); +} + +static void +flac_decode(struct decoder * decoder, struct input_stream *input_stream) +{ + flac_decode_internal(decoder, input_stream, false); +} + +#ifndef HAVE_OGGFLAC + +static bool +oggflac_init(G_GNUC_UNUSED const struct config_param *param) +{ +#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7 + return !!FLAC_API_SUPPORTS_OGG_FLAC; +#else + /* disable oggflac when libflac is too old */ + return false; +#endif +} + +#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7 + +static struct tag * +oggflac_tag_dup(const char *file) +{ + struct tag *ret = NULL; + FLAC__Metadata_Iterator *it; + FLAC__StreamMetadata *block; + FLAC__Metadata_Chain *chain = FLAC__metadata_chain_new(); + + if (!(FLAC__metadata_chain_read_ogg(chain, file))) + goto out; + it = FLAC__metadata_iterator_new(); + FLAC__metadata_iterator_init(it, chain); + + ret = tag_new(); + do { + if (!(block = FLAC__metadata_iterator_get_block(it))) + break; + + flac_tag_apply_metadata(ret, NULL, block); + } while (FLAC__metadata_iterator_next(it)); + FLAC__metadata_iterator_delete(it); + + if (!tag_is_defined(ret)) { + tag_free(ret); + ret = NULL; + } + +out: + FLAC__metadata_chain_delete(chain); + return ret; +} + +static void +oggflac_decode(struct decoder *decoder, struct input_stream *input_stream) +{ + if (ogg_stream_type_detect(input_stream) != FLAC) + return; + + /* rewind the stream, because ogg_stream_type_detect() has + moved it */ + input_stream_seek(input_stream, 0, SEEK_SET, NULL); + + flac_decode_internal(decoder, input_stream, true); +} + +static const char *const oggflac_suffixes[] = { "ogg", "oga", NULL }; +static const char *const oggflac_mime_types[] = { + "application/ogg", + "application/x-ogg", + "audio/ogg", + "audio/x-flac+ogg", + "audio/x-ogg", + NULL +}; + +#endif /* FLAC_API_VERSION_CURRENT >= 7 */ + +const struct decoder_plugin oggflac_decoder_plugin = { + .name = "oggflac", + .init = oggflac_init, +#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7 + .stream_decode = oggflac_decode, + .tag_dup = oggflac_tag_dup, + .suffixes = oggflac_suffixes, + .mime_types = oggflac_mime_types +#endif +}; + +#endif /* HAVE_OGGFLAC */ + +static const char *const flac_suffixes[] = { "flac", NULL }; +static const char *const flac_mime_types[] = { + "application/flac", + "application/x-flac", + "audio/flac", + "audio/x-flac", + NULL +}; + +const struct decoder_plugin flac_decoder_plugin = { + .name = "flac", + .stream_decode = flac_decode, + .tag_dup = flac_tag_dup, + .suffixes = flac_suffixes, + .mime_types = flac_mime_types, +}; diff --git a/src/decoder/flac_metadata.c b/src/decoder/flac_metadata.c new file mode 100644 index 000000000..f2f2f954d --- /dev/null +++ b/src/decoder/flac_metadata.c @@ -0,0 +1,286 @@ +/* + * Copyright (C) 2003-2010 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 "flac_metadata.h" +#include "replay_gain_info.h" +#include "tag.h" + +#include <glib.h> + +#include <assert.h> +#include <stdbool.h> +#include <stdlib.h> + +static bool +flac_find_float_comment(const FLAC__StreamMetadata *block, + const char *cmnt, float *fl) +{ + int offset; + size_t pos; + int len; + unsigned char tmp, *p; + + offset = FLAC__metadata_object_vorbiscomment_find_entry_from(block, 0, + cmnt); + if (offset < 0) + return false; + + pos = strlen(cmnt) + 1; /* 1 is for '=' */ + len = block->data.vorbis_comment.comments[offset].length - pos; + if (len <= 0) + return false; + + p = &block->data.vorbis_comment.comments[offset].entry[pos]; + tmp = p[len]; + p[len] = '\0'; + *fl = (float)atof((char *)p); + p[len] = tmp; + + return true; +} + +bool +flac_parse_replay_gain(struct replay_gain_info *rgi, + const FLAC__StreamMetadata *block) +{ + bool found = false; + + replay_gain_info_init(rgi); + + if (flac_find_float_comment(block, "replaygain_album_gain", + &rgi->tuples[REPLAY_GAIN_ALBUM].gain)) + found = true; + if (flac_find_float_comment(block, "replaygain_album_peak", + &rgi->tuples[REPLAY_GAIN_ALBUM].peak)) + found = true; + if (flac_find_float_comment(block, "replaygain_track_gain", + &rgi->tuples[REPLAY_GAIN_TRACK].gain)) + found = true; + if (flac_find_float_comment(block, "replaygain_track_peak", + &rgi->tuples[REPLAY_GAIN_TRACK].peak)) + found = true; + + return found; +} + +static bool +flac_find_string_comment(const FLAC__StreamMetadata *block, + const char *cmnt, char **str) +{ + int offset; + size_t pos; + int len; + const unsigned char *p; + + *str = NULL; + offset = FLAC__metadata_object_vorbiscomment_find_entry_from(block, 0, + cmnt); + if (offset < 0) + return false; + + pos = strlen(cmnt) + 1; /* 1 is for '=' */ + len = block->data.vorbis_comment.comments[offset].length - pos; + if (len <= 0) + return false; + + p = &block->data.vorbis_comment.comments[offset].entry[pos]; + *str = g_strndup((const char *)p, len); + + return true; +} + +bool +flac_parse_mixramp(char **mixramp_start, char **mixramp_end, + const FLAC__StreamMetadata *block) +{ + bool found = false; + + if (flac_find_string_comment(block, "mixramp_start", mixramp_start)) + found = true; + if (flac_find_string_comment(block, "mixramp_end", mixramp_end)) + found = true; + + return found; +} + +/** + * Checks if the specified name matches the entry's name, and if yes, + * returns the comment value (not null-temrinated). + */ +static const char * +flac_comment_value(const FLAC__StreamMetadata_VorbisComment_Entry *entry, + const char *name, const char *char_tnum, size_t *length_r) +{ + size_t name_length = strlen(name); + size_t char_tnum_length = 0; + const char *comment = (const char*)entry->entry; + + if (entry->length <= name_length || + g_ascii_strncasecmp(comment, name, name_length) != 0) + return NULL; + + if (char_tnum != NULL) { + char_tnum_length = strlen(char_tnum); + if (entry->length > name_length + char_tnum_length + 2 && + comment[name_length] == '[' && + g_ascii_strncasecmp(comment + name_length + 1, + char_tnum, char_tnum_length) == 0 && + comment[name_length + char_tnum_length + 1] == ']') + name_length = name_length + char_tnum_length + 2; + else if (entry->length > name_length + char_tnum_length && + g_ascii_strncasecmp(comment + name_length, + char_tnum, char_tnum_length) == 0) + name_length = name_length + char_tnum_length; + } + + if (comment[name_length] == '=') { + *length_r = entry->length - name_length - 1; + return comment + name_length + 1; + } + + return NULL; +} + +/** + * Check if the comment's name equals the passed name, and if so, copy + * the comment value into the tag. + */ +static bool +flac_copy_comment(struct tag *tag, + const FLAC__StreamMetadata_VorbisComment_Entry *entry, + const char *name, enum tag_type tag_type, + const char *char_tnum) +{ + const char *value; + size_t value_length; + + value = flac_comment_value(entry, name, char_tnum, &value_length); + if (value != NULL) { + tag_add_item_n(tag, tag_type, value, value_length); + return true; + } + + return false; +} + +/* tracknumber is used in VCs, MPD uses "track" ..., all the other + * tag names match */ +static const char *VORBIS_COMMENT_TRACK_KEY = "tracknumber"; +static const char *VORBIS_COMMENT_DISC_KEY = "discnumber"; + +static void +flac_parse_comment(struct tag *tag, const char *char_tnum, + const FLAC__StreamMetadata_VorbisComment_Entry *entry) +{ + assert(tag != NULL); + + if (flac_copy_comment(tag, entry, VORBIS_COMMENT_TRACK_KEY, + TAG_TRACK, char_tnum) || + flac_copy_comment(tag, entry, VORBIS_COMMENT_DISC_KEY, + TAG_DISC, char_tnum) || + flac_copy_comment(tag, entry, "album artist", + TAG_ALBUM_ARTIST, char_tnum)) + return; + + for (unsigned i = 0; i < TAG_NUM_OF_ITEM_TYPES; ++i) + if (flac_copy_comment(tag, entry, + tag_item_names[i], i, char_tnum)) + return; +} + +void +flac_vorbis_comments_to_tag(struct tag *tag, const char *char_tnum, + const FLAC__StreamMetadata_VorbisComment *comment) +{ + for (unsigned i = 0; i < comment->num_comments; ++i) + flac_parse_comment(tag, char_tnum, &comment->comments[i]); +} + +void +flac_tag_apply_metadata(struct tag *tag, const char *track, + const FLAC__StreamMetadata *block) +{ + switch (block->type) { + case FLAC__METADATA_TYPE_VORBIS_COMMENT: + flac_vorbis_comments_to_tag(tag, track, + &block->data.vorbis_comment); + break; + + case FLAC__METADATA_TYPE_STREAMINFO: + tag->time = flac_duration(&block->data.stream_info); + break; + + default: + break; + } +} + +struct tag * +flac_tag_load(const char *file, const char *char_tnum) +{ + struct tag *tag; + FLAC__Metadata_SimpleIterator *it; + FLAC__StreamMetadata *block = NULL; + + it = FLAC__metadata_simple_iterator_new(); + if (!FLAC__metadata_simple_iterator_init(it, file, 1, 0)) { + const char *err; + FLAC_API FLAC__Metadata_SimpleIteratorStatus s; + + s = FLAC__metadata_simple_iterator_status(it); + + switch (s) { /* slightly more human-friendly messages: */ + case FLAC__METADATA_SIMPLE_ITERATOR_STATUS_ILLEGAL_INPUT: + err = "illegal input"; + break; + case FLAC__METADATA_SIMPLE_ITERATOR_STATUS_ERROR_OPENING_FILE: + err = "error opening file"; + break; + case FLAC__METADATA_SIMPLE_ITERATOR_STATUS_NOT_A_FLAC_FILE: + err = "not a FLAC file"; + break; + default: + err = FLAC__Metadata_SimpleIteratorStatusString[s]; + } + g_debug("Reading '%s' metadata gave the following error: %s\n", + file, err); + FLAC__metadata_simple_iterator_delete(it); + return NULL; + } + + tag = tag_new(); + do { + block = FLAC__metadata_simple_iterator_get_block(it); + if (!block) + break; + + flac_tag_apply_metadata(tag, char_tnum, block); + FLAC__metadata_object_delete(block); + } while (FLAC__metadata_simple_iterator_next(it)); + + FLAC__metadata_simple_iterator_delete(it); + + if (!tag_is_defined(tag)) { + tag_free(tag); + tag = NULL; + } + + return tag; +} diff --git a/src/decoder/flac_metadata.h b/src/decoder/flac_metadata.h new file mode 100644 index 000000000..06e691d1d --- /dev/null +++ b/src/decoder/flac_metadata.h @@ -0,0 +1,55 @@ +/* + * Copyright (C) 2003-2010 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_FLAC_METADATA_H +#define MPD_FLAC_METADATA_H + +#include <stdbool.h> +#include <FLAC/metadata.h> + +struct tag; +struct replay_gain_info; + +static inline unsigned +flac_duration(const FLAC__StreamMetadata_StreamInfo *stream_info) +{ + return (stream_info->total_samples + stream_info->sample_rate - 1) / + stream_info->sample_rate; +} + +bool +flac_parse_replay_gain(struct replay_gain_info *rgi, + const FLAC__StreamMetadata *block); + +bool +flac_parse_mixramp(char **mixramp_start, char **mixramp_end, + const FLAC__StreamMetadata *block); + +void +flac_vorbis_comments_to_tag(struct tag *tag, const char *char_tnum, + const FLAC__StreamMetadata_VorbisComment *comment); + +void +flac_tag_apply_metadata(struct tag *tag, const char *track, + const FLAC__StreamMetadata *block); + +struct tag * +flac_tag_load(const char *file, const char *char_tnum); + +#endif diff --git a/src/decoder/flac_pcm.c b/src/decoder/flac_pcm.c new file mode 100644 index 000000000..bf6e2612c --- /dev/null +++ b/src/decoder/flac_pcm.c @@ -0,0 +1,109 @@ +/* + * Copyright (C) 2003-2010 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 "flac_pcm.h" + +#include <assert.h> + +static void flac_convert_stereo16(int16_t *dest, + const FLAC__int32 * const buf[], + unsigned int position, unsigned int end) +{ + for (; position < end; ++position) { + *dest++ = buf[0][position]; + *dest++ = buf[1][position]; + } +} + +static void +flac_convert_16(int16_t *dest, + unsigned int num_channels, + const FLAC__int32 * const buf[], + unsigned int position, unsigned int end) +{ + unsigned int c_chan; + + for (; position < end; ++position) + for (c_chan = 0; c_chan < num_channels; c_chan++) + *dest++ = buf[c_chan][position]; +} + +/** + * Note: this function also handles 24 bit files! + */ +static void +flac_convert_32(int32_t *dest, + unsigned int num_channels, + const FLAC__int32 * const buf[], + unsigned int position, unsigned int end) +{ + unsigned int c_chan; + + for (; position < end; ++position) + for (c_chan = 0; c_chan < num_channels; c_chan++) + *dest++ = buf[c_chan][position]; +} + +static void +flac_convert_8(int8_t *dest, + unsigned int num_channels, + const FLAC__int32 * const buf[], + unsigned int position, unsigned int end) +{ + unsigned int c_chan; + + for (; position < end; ++position) + for (c_chan = 0; c_chan < num_channels; c_chan++) + *dest++ = buf[c_chan][position]; +} + +void +flac_convert(void *dest, + unsigned int num_channels, enum sample_format sample_format, + const FLAC__int32 *const buf[], + unsigned int position, unsigned int end) +{ + switch (sample_format) { + case SAMPLE_FORMAT_S16: + if (num_channels == 2) + flac_convert_stereo16((int16_t*)dest, buf, + position, end); + else + flac_convert_16((int16_t*)dest, num_channels, buf, + position, end); + break; + + case SAMPLE_FORMAT_S24_P32: + case SAMPLE_FORMAT_S32: + flac_convert_32((int32_t*)dest, num_channels, buf, + position, end); + break; + + case SAMPLE_FORMAT_S8: + flac_convert_8((int8_t*)dest, num_channels, buf, + position, end); + break; + + case SAMPLE_FORMAT_S24: + case SAMPLE_FORMAT_UNDEFINED: + /* unreachable */ + assert(false); + } +} diff --git a/src/decoder/flac_pcm.h b/src/decoder/flac_pcm.h new file mode 100644 index 000000000..bccfc645c --- /dev/null +++ b/src/decoder/flac_pcm.h @@ -0,0 +1,33 @@ +/* + * Copyright (C) 2003-2010 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_FLAC_PCM_H +#define MPD_FLAC_PCM_H + +#include "audio_format.h" + +#include <FLAC/ordinals.h> + +void +flac_convert(void *dest, + unsigned int num_channels, enum sample_format sample_format, + const FLAC__int32 *const buf[], + unsigned int position, unsigned int end); + +#endif diff --git a/src/decoder/flac_plugin.c b/src/decoder/flac_plugin.c deleted file mode 100644 index 1e568f70d..000000000 --- a/src/decoder/flac_plugin.c +++ /dev/null @@ -1,918 +0,0 @@ -/* - * 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 "_flac_common.h" - -#include <glib.h> - -#include <assert.h> -#include <unistd.h> - -#include <sys/stat.h> -#include <sys/types.h> - -#ifdef HAVE_CUE /* libcue */ -#include "../cue/cue_tag.h" -#endif - -/* this code was based on flac123, from flac-tools */ - -static flac_read_status -flac_read_cb(G_GNUC_UNUSED const flac_decoder *fd, - FLAC__byte buf[], flac_read_status_size_t *bytes, - void *fdata) -{ - struct flac_data *data = fdata; - size_t r; - - r = decoder_read(data->decoder, data->input_stream, - (void *)buf, *bytes); - *bytes = r; - - if (r == 0) { - if (decoder_get_command(data->decoder) != DECODE_COMMAND_NONE || - input_stream_eof(data->input_stream)) - return flac_read_status_eof; - else - return flac_read_status_abort; - } - - return flac_read_status_continue; -} - -static flac_seek_status -flac_seek_cb(G_GNUC_UNUSED const flac_decoder *fd, - FLAC__uint64 offset, void *fdata) -{ - struct flac_data *data = (struct flac_data *) fdata; - - if (!input_stream_seek(data->input_stream, offset, SEEK_SET)) - return flac_seek_status_error; - - return flac_seek_status_ok; -} - -static flac_tell_status -flac_tell_cb(G_GNUC_UNUSED const flac_decoder *fd, - FLAC__uint64 * offset, void *fdata) -{ - struct flac_data *data = (struct flac_data *) fdata; - - *offset = (long)(data->input_stream->offset); - - return flac_tell_status_ok; -} - -static flac_length_status -flac_length_cb(G_GNUC_UNUSED const flac_decoder *fd, - FLAC__uint64 * length, void *fdata) -{ - struct flac_data *data = (struct flac_data *) fdata; - - if (data->input_stream->size < 0) - return flac_length_status_unsupported; - - *length = (size_t) (data->input_stream->size); - - return flac_length_status_ok; -} - -static FLAC__bool -flac_eof_cb(G_GNUC_UNUSED const flac_decoder *fd, void *fdata) -{ - struct flac_data *data = (struct flac_data *) fdata; - - return (decoder_get_command(data->decoder) != DECODE_COMMAND_NONE && - decoder_get_command(data->decoder) != DECODE_COMMAND_SEEK) || - input_stream_eof(data->input_stream); -} - -static void -flac_error_cb(G_GNUC_UNUSED const flac_decoder *fd, - FLAC__StreamDecoderErrorStatus status, void *fdata) -{ - flac_error_common_cb("flac", status, (struct flac_data *) fdata); -} - -#if !defined(FLAC_API_VERSION_CURRENT) || FLAC_API_VERSION_CURRENT <= 7 -static void flacPrintErroredState(FLAC__SeekableStreamDecoderState state) -{ - const char *str = ""; /* "" to silence compiler warning */ - switch (state) { - case FLAC__SEEKABLE_STREAM_DECODER_OK: - case FLAC__SEEKABLE_STREAM_DECODER_SEEKING: - case FLAC__SEEKABLE_STREAM_DECODER_END_OF_STREAM: - return; - case FLAC__SEEKABLE_STREAM_DECODER_MEMORY_ALLOCATION_ERROR: - str = "allocation error"; - break; - case FLAC__SEEKABLE_STREAM_DECODER_READ_ERROR: - str = "read error"; - break; - case FLAC__SEEKABLE_STREAM_DECODER_SEEK_ERROR: - str = "seek error"; - break; - case FLAC__SEEKABLE_STREAM_DECODER_STREAM_DECODER_ERROR: - str = "seekable stream error"; - break; - case FLAC__SEEKABLE_STREAM_DECODER_ALREADY_INITIALIZED: - str = "decoder already initialized"; - break; - case FLAC__SEEKABLE_STREAM_DECODER_INVALID_CALLBACK: - str = "invalid callback"; - break; - case FLAC__SEEKABLE_STREAM_DECODER_UNINITIALIZED: - str = "decoder uninitialized"; - } - - g_warning("%s\n", str); -} - -static bool -flac_init(FLAC__SeekableStreamDecoder *dec, - FLAC__SeekableStreamDecoderReadCallback read_cb, - FLAC__SeekableStreamDecoderSeekCallback seek_cb, - FLAC__SeekableStreamDecoderTellCallback tell_cb, - FLAC__SeekableStreamDecoderLengthCallback length_cb, - FLAC__SeekableStreamDecoderEofCallback eof_cb, - FLAC__SeekableStreamDecoderWriteCallback write_cb, - FLAC__SeekableStreamDecoderMetadataCallback metadata_cb, - FLAC__SeekableStreamDecoderErrorCallback error_cb, - void *data) -{ - return FLAC__seekable_stream_decoder_set_read_callback(dec, read_cb) && - FLAC__seekable_stream_decoder_set_seek_callback(dec, seek_cb) && - FLAC__seekable_stream_decoder_set_tell_callback(dec, tell_cb) && - FLAC__seekable_stream_decoder_set_length_callback(dec, length_cb) && - FLAC__seekable_stream_decoder_set_eof_callback(dec, eof_cb) && - FLAC__seekable_stream_decoder_set_write_callback(dec, write_cb) && - FLAC__seekable_stream_decoder_set_metadata_callback(dec, metadata_cb) && - FLAC__seekable_stream_decoder_set_metadata_respond(dec, FLAC__METADATA_TYPE_VORBIS_COMMENT) && - FLAC__seekable_stream_decoder_set_error_callback(dec, error_cb) && - FLAC__seekable_stream_decoder_set_client_data(dec, data) && - FLAC__seekable_stream_decoder_init(dec) == FLAC__SEEKABLE_STREAM_DECODER_OK; -} -#else /* FLAC_API_VERSION_CURRENT >= 7 */ -static void flacPrintErroredState(FLAC__StreamDecoderState state) -{ - const char *str = ""; /* "" to silence compiler warning */ - switch (state) { - case FLAC__STREAM_DECODER_SEARCH_FOR_METADATA: - case FLAC__STREAM_DECODER_READ_METADATA: - case FLAC__STREAM_DECODER_SEARCH_FOR_FRAME_SYNC: - case FLAC__STREAM_DECODER_READ_FRAME: - case FLAC__STREAM_DECODER_END_OF_STREAM: - return; - case FLAC__STREAM_DECODER_OGG_ERROR: - str = "error in the Ogg layer"; - break; - case FLAC__STREAM_DECODER_SEEK_ERROR: - str = "seek error"; - break; - case FLAC__STREAM_DECODER_ABORTED: - str = "decoder aborted by read"; - break; - case FLAC__STREAM_DECODER_MEMORY_ALLOCATION_ERROR: - str = "allocation error"; - break; - case FLAC__STREAM_DECODER_UNINITIALIZED: - str = "decoder uninitialized"; - } - - g_warning("%s\n", str); -} -#endif /* FLAC_API_VERSION_CURRENT >= 7 */ - -static void flacMetadata(G_GNUC_UNUSED const flac_decoder * dec, - const FLAC__StreamMetadata * block, void *vdata) -{ - flac_metadata_common_cb(block, (struct flac_data *) vdata); -} - -static FLAC__StreamDecoderWriteStatus -flac_write_cb(const flac_decoder *dec, const FLAC__Frame *frame, - const FLAC__int32 *const buf[], void *vdata) -{ - FLAC__uint32 samples = frame->header.blocksize; - struct flac_data *data = (struct flac_data *) vdata; - float timeChange; - FLAC__uint64 newPosition = 0; - - timeChange = ((float)samples) / frame->header.sample_rate; - data->time += timeChange; - - flac_get_decode_position(dec, &newPosition); - if (data->position && newPosition >= data->position) { - assert(timeChange >= 0); - - data->bit_rate = - ((newPosition - data->position) * 8.0 / timeChange) - / 1000 + 0.5; - } - data->position = newPosition; - - return flac_common_write(data, frame, buf); -} - -static struct tag * -flac_tag_load(const char *file, const char *char_tnum) -{ - struct tag *tag; - FLAC__Metadata_SimpleIterator *it; - FLAC__StreamMetadata *block = NULL; - - it = FLAC__metadata_simple_iterator_new(); - if (!FLAC__metadata_simple_iterator_init(it, file, 1, 0)) { - const char *err; - FLAC_API FLAC__Metadata_SimpleIteratorStatus s; - - s = FLAC__metadata_simple_iterator_status(it); - - switch (s) { /* slightly more human-friendly messages: */ - case FLAC__METADATA_SIMPLE_ITERATOR_STATUS_ILLEGAL_INPUT: - err = "illegal input"; - break; - case FLAC__METADATA_SIMPLE_ITERATOR_STATUS_ERROR_OPENING_FILE: - err = "error opening file"; - break; - case FLAC__METADATA_SIMPLE_ITERATOR_STATUS_NOT_A_FLAC_FILE: - err = "not a FLAC file"; - break; - default: - err = FLAC__Metadata_SimpleIteratorStatusString[s]; - } - g_debug("Reading '%s' metadata gave the following error: %s\n", - file, err); - FLAC__metadata_simple_iterator_delete(it); - return NULL; - } - - tag = tag_new(); - do { - block = FLAC__metadata_simple_iterator_get_block(it); - if (!block) - break; - if (block->type == FLAC__METADATA_TYPE_VORBIS_COMMENT) { - flac_vorbis_comments_to_tag(tag, char_tnum, block); - } else if (block->type == FLAC__METADATA_TYPE_STREAMINFO) { - tag->time = ((float)block->data.stream_info.total_samples) / - block->data.stream_info.sample_rate + 0.5; - } - FLAC__metadata_object_delete(block); - } while (FLAC__metadata_simple_iterator_next(it)); - - FLAC__metadata_simple_iterator_delete(it); - - if (!tag_is_defined(tag)) { - tag_free(tag); - tag = NULL; - } - - return tag; -} - -#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7 - -static struct tag * -flac_cue_tag_load(const char *file) -{ - struct tag* tag = NULL; - char* char_tnum = NULL; - char* ptr = NULL; - unsigned int tnum = 0; - unsigned int sample_rate = 0; - FLAC__uint64 track_time = 0; -#ifdef HAVE_CUE /* libcue */ - FLAC__StreamMetadata* vc; -#endif /* libcue */ - FLAC__StreamMetadata* si = FLAC__metadata_object_new(FLAC__METADATA_TYPE_STREAMINFO); - FLAC__StreamMetadata* cs; - - tnum = flac_vtrack_tnum(file); - char_tnum = g_strdup_printf("%u", tnum); - - ptr = strrchr(file, '/'); - *ptr = '\0'; - -#ifdef HAVE_CUE /* libcue */ - if (FLAC__metadata_get_tags(file, &vc)) - { - for (unsigned i = 0; i < vc->data.vorbis_comment.num_comments; - i++) - { - if ((ptr = (char*)vc->data.vorbis_comment.comments[i].entry) != NULL) - { - if (g_ascii_strncasecmp(ptr, "cuesheet", 8) == 0) - { - while (*(++ptr) != '='); - tag = cue_tag_string( ++ptr, - tnum); - } - } - } - - FLAC__metadata_object_delete(vc); - } -#endif /* libcue */ - - if (tag == NULL) - tag = flac_tag_load(file, char_tnum); - - if (char_tnum != NULL) - { - tag_add_item( tag, - TAG_ITEM_TRACK, - char_tnum); - g_free(char_tnum); - } - - if (FLAC__metadata_get_streaminfo(file, si)) - { - sample_rate = si->data.stream_info.sample_rate; - FLAC__metadata_object_delete(si); - } - - if (FLAC__metadata_get_cuesheet(file, &cs)) - { - if (cs->data.cue_sheet.tracks != NULL - && (tnum <= cs->data.cue_sheet.num_tracks - 1)) - { - track_time = cs->data.cue_sheet.tracks[tnum].offset - - cs->data.cue_sheet.tracks[tnum - 1].offset; - } - FLAC__metadata_object_delete(cs); - } - - if (sample_rate != 0) - { - tag->time = (unsigned int)(track_time/sample_rate); - } - - return tag; -} - -#endif /* FLAC_API_VERSION_CURRENT >= 7 */ - -static struct tag * -flac_tag_dup(const char *file) -{ -#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7 - struct stat st; - - if (stat(file, &st) < 0) - return flac_cue_tag_load(file); - else -#endif /* FLAC_API_VERSION_CURRENT >= 7 */ - return flac_tag_load(file, NULL); -} - -static void -flac_decode_internal(struct decoder * decoder, - struct input_stream *input_stream, - bool is_ogg) -{ - flac_decoder *flac_dec; - struct flac_data data; - enum decoder_command cmd; - const char *err = NULL; - - if (!(flac_dec = flac_new())) - return; - flac_data_init(&data, decoder, input_stream); - data.tag = tag_new(); - -#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7 - if(!FLAC__stream_decoder_set_metadata_respond(flac_dec, FLAC__METADATA_TYPE_VORBIS_COMMENT)) - { - g_debug("Failed to set metadata respond\n"); - } -#endif - - if (is_ogg) { - if (!flac_ogg_init(flac_dec, flac_read_cb, - flac_seek_cb, flac_tell_cb, - flac_length_cb, flac_eof_cb, - flac_write_cb, flacMetadata, - flac_error_cb, (void *)&data)) { - err = "doing Ogg init()"; - goto fail; - } - } else { - if (!flac_init(flac_dec, flac_read_cb, - flac_seek_cb, flac_tell_cb, - flac_length_cb, flac_eof_cb, - flac_write_cb, flacMetadata, - flac_error_cb, (void *)&data)) { - err = "doing init()"; - goto fail; - } - } - - if (!flac_process_metadata(flac_dec)) { - err = "problem reading metadata"; - goto fail; - } - - if (!audio_format_valid(&data.audio_format)) { - g_warning("Invalid audio format: %u:%u:%u\n", - data.audio_format.sample_rate, - data.audio_format.bits, - data.audio_format.channels); - goto fail; - } - - decoder_initialized(decoder, &data.audio_format, - input_stream->seekable, data.total_time); - - while (true) { - if (!tag_is_empty(data.tag)) { - cmd = decoder_tag(decoder, input_stream, data.tag); - tag_free(data.tag); - data.tag = tag_new(); - } else - cmd = decoder_get_command(decoder); - - if (cmd == DECODE_COMMAND_SEEK) { - FLAC__uint64 seek_sample = decoder_seek_where(decoder) * - data.audio_format.sample_rate + 0.5; - if (flac_seek_absolute(flac_dec, seek_sample)) { - data.time = ((float)seek_sample) / - data.audio_format.sample_rate; - data.position = 0; - decoder_command_finished(decoder); - } else - decoder_seek_error(decoder); - } else if (cmd == DECODE_COMMAND_STOP || - flac_get_state(flac_dec) == flac_decoder_eof) - break; - - if (!flac_process_single(flac_dec)) { - cmd = decoder_get_command(decoder); - if (cmd != DECODE_COMMAND_SEEK) - break; - } - } - if (cmd != DECODE_COMMAND_STOP) { - flacPrintErroredState(flac_get_state(flac_dec)); - flac_finish(flac_dec); - } - -fail: - if (data.replay_gain_info) - replay_gain_info_free(data.replay_gain_info); - - tag_free(data.tag); - - if (flac_dec) - flac_delete(flac_dec); - - if (err) - g_warning("%s\n", err); -} - -static void -flac_decode(struct decoder * decoder, struct input_stream *input_stream) -{ - flac_decode_internal(decoder, input_stream, false); -} - -#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7 - -/** - * @brief Decode a flac file with embedded cue sheets - * @param const char* fname filename on fs - */ -static void -flac_container_decode(struct decoder* decoder, - const char* fname, - bool is_ogg) -{ - unsigned int tnum = 0; - FLAC__uint64 t_start = 0; - FLAC__uint64 t_end = 0; - FLAC__uint64 track_time = 0; - FLAC__StreamMetadata* cs = NULL; - - flac_decoder *flac_dec; - struct flac_data data; - const char *err = NULL; - - char* pathname = g_strdup(fname); - char* slash = strrchr(pathname, '/'); - *slash = '\0'; - - tnum = flac_vtrack_tnum(fname); - - cs = FLAC__metadata_object_new(FLAC__METADATA_TYPE_CUESHEET); - - FLAC__metadata_get_cuesheet(pathname, &cs); - - if (cs != NULL) - { - if (cs->data.cue_sheet.tracks != NULL - && (tnum <= cs->data.cue_sheet.num_tracks - 1)) - { - t_start = cs->data.cue_sheet.tracks[tnum - 1].offset; - t_end = cs->data.cue_sheet.tracks[tnum].offset; - track_time = cs->data.cue_sheet.tracks[tnum].offset - - cs->data.cue_sheet.tracks[tnum - 1].offset; - } - - FLAC__metadata_object_delete(cs); - } - else - { - g_free(pathname); - return; - } - - if (!(flac_dec = flac_new())) - { - g_free(pathname); - return; - } - - flac_data_init(&data, decoder, NULL); - -#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7 - if(!FLAC__stream_decoder_set_metadata_respond(flac_dec, FLAC__METADATA_TYPE_VORBIS_COMMENT)) - { - g_debug("Failed to set metadata respond\n"); - } -#endif - - - if (is_ogg) - { - if (FLAC__stream_decoder_init_ogg_file( flac_dec, - pathname, - flac_write_cb, - flacMetadata, - flac_error_cb, - (void*) &data ) - != FLAC__STREAM_DECODER_INIT_STATUS_OK ) - { - err = "doing Ogg init()"; - goto fail; - } - } - else - { - if (FLAC__stream_decoder_init_file( flac_dec, - pathname, - flac_write_cb, - flacMetadata, - flac_error_cb, - (void*) &data ) - != FLAC__STREAM_DECODER_INIT_STATUS_OK ) - { - err = "doing init()"; - goto fail; - } - } - - if (!flac_process_metadata(flac_dec)) - { - err = "problem reading metadata"; - goto fail; - } - - if (!audio_format_valid(&data.audio_format)) - { - g_warning("Invalid audio format: %u:%u:%u\n", - data.audio_format.sample_rate, - data.audio_format.bits, - data.audio_format.channels); - goto fail; - } - - // set track time (order is important: after stream init) - data.total_time = ((float)track_time / (float)data.audio_format.sample_rate); - data.position = 0; - - decoder_initialized(decoder, &data.audio_format, - true, data.total_time); - - // seek to song start (order is important: after decoder init) - flac_seek_absolute(flac_dec, (FLAC__uint64)t_start); - - while (true) - { - if (!flac_process_single(flac_dec)) - break; - - // we only need to break at the end of track if we are in "cue mode" - if (data.time >= data.total_time) - { - flacPrintErroredState(flac_get_state(flac_dec)); - flac_finish(flac_dec); - } - - if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) - { - FLAC__uint64 seek_sample = t_start + - (decoder_seek_where(decoder) * data.audio_format.sample_rate); - - if (seek_sample >= t_start && seek_sample <= t_end && - flac_seek_absolute(flac_dec, (FLAC__uint64)seek_sample)) { - data.time = (float)(seek_sample - t_start) / - data.audio_format.sample_rate; - data.position = 0; - - decoder_command_finished(decoder); - } else - decoder_seek_error(decoder); - } - else if (flac_get_state(flac_dec) == flac_decoder_eof) - break; - } - - if (decoder_get_command(decoder) != DECODE_COMMAND_STOP) - { - flacPrintErroredState(flac_get_state(flac_dec)); - flac_finish(flac_dec); - } - -fail: - if (pathname) - g_free(pathname); - - if (data.replay_gain_info) - replay_gain_info_free(data.replay_gain_info); - - if (flac_dec) - flac_delete(flac_dec); - - if (err) - g_warning("%s\n", err); -} - -/** - * @brief Open a flac file for decoding - * @param const char* fname filename on fs - */ -static void -flac_filedecode_internal(struct decoder* decoder, - const char* fname, - bool is_ogg) -{ - flac_decoder *flac_dec; - struct flac_data data; - const char *err = NULL; - unsigned int flac_err_state = 0; - - if (!(flac_dec = flac_new())) - return; - - flac_data_init(&data, decoder, NULL); - -#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7 - if(!FLAC__stream_decoder_set_metadata_respond(flac_dec, FLAC__METADATA_TYPE_VORBIS_COMMENT)) - { - g_debug("Failed to set metadata respond\n"); - } -#endif - - - if (is_ogg) - { - if ( (flac_err_state = FLAC__stream_decoder_init_ogg_file( flac_dec, - fname, - flac_write_cb, - flacMetadata, - flac_error_cb, - (void*) &data )) - == FLAC__STREAM_DECODER_INIT_STATUS_ERROR_OPENING_FILE) - { - flac_container_decode(decoder, fname, is_ogg); - } - else if (flac_err_state != FLAC__STREAM_DECODER_INIT_STATUS_OK) - { - err = "doing Ogg init()"; - goto fail; - } - } - else - { - if ( (flac_err_state = FLAC__stream_decoder_init_file( flac_dec, - fname, - flac_write_cb, - flacMetadata, - flac_error_cb, - (void*) &data )) - == FLAC__STREAM_DECODER_INIT_STATUS_ERROR_OPENING_FILE) - { - flac_container_decode(decoder, fname, is_ogg); - } - else if (flac_err_state != FLAC__STREAM_DECODER_INIT_STATUS_OK) - { - err = "doing init()"; - goto fail; - } - } - - if (!flac_process_metadata(flac_dec)) - { - err = "problem reading metadata"; - goto fail; - } - - if (!audio_format_valid(&data.audio_format)) - { - g_warning("Invalid audio format: %u:%u:%u\n", - data.audio_format.sample_rate, - data.audio_format.bits, - data.audio_format.channels); - goto fail; - } - - decoder_initialized(decoder, &data.audio_format, - true, data.total_time); - - while (true) - { - if (!flac_process_single(flac_dec)) - break; - - if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) - { - FLAC__uint64 seek_sample = decoder_seek_where(decoder) * - data.audio_format.sample_rate + 0.5; - if (flac_seek_absolute(flac_dec, seek_sample)) - { - data.time = ((float)seek_sample) / - data.audio_format.sample_rate; - data.position = 0; - decoder_command_finished(decoder); - } - else - decoder_seek_error(decoder); - - } - else if (flac_get_state(flac_dec) == flac_decoder_eof) - break; - } - - if (decoder_get_command(decoder) != DECODE_COMMAND_STOP) - { - flacPrintErroredState(flac_get_state(flac_dec)); - flac_finish(flac_dec); - } - -fail: - if (data.replay_gain_info) - replay_gain_info_free(data.replay_gain_info); - - if (flac_dec) - flac_delete(flac_dec); - - if (err) - g_warning("%s\n", err); -} - -/** - * @brief wrapper function for - * flac_filedecode_internal method - * for decoding without ogg - */ -static void -flac_filedecode(struct decoder *decoder, const char *fname) -{ - struct stat st; - - if (stat(fname, &st) < 0) { - flac_container_decode(decoder, fname, false); - } else - flac_filedecode_internal(decoder, fname, false); -} - -#endif /* FLAC_API_VERSION_CURRENT >= 7 */ - -#ifndef HAVE_OGGFLAC - -static bool -oggflac_init(G_GNUC_UNUSED const struct config_param *param) -{ -#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7 - return !!FLAC_API_SUPPORTS_OGG_FLAC; -#else - /* disable oggflac when libflac is too old */ - return false; -#endif -} - -#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7 - -static struct tag * -oggflac_tag_dup(const char *file) -{ - struct tag *ret = NULL; - FLAC__Metadata_Iterator *it; - FLAC__StreamMetadata *block; - FLAC__Metadata_Chain *chain = FLAC__metadata_chain_new(); - - if (!(FLAC__metadata_chain_read_ogg(chain, file))) - goto out; - it = FLAC__metadata_iterator_new(); - FLAC__metadata_iterator_init(it, chain); - - ret = tag_new(); - do { - if (!(block = FLAC__metadata_iterator_get_block(it))) - break; - if (block->type == FLAC__METADATA_TYPE_VORBIS_COMMENT) { - flac_vorbis_comments_to_tag(ret, NULL, block); - } else if (block->type == FLAC__METADATA_TYPE_STREAMINFO) { - ret->time = ((float)block->data.stream_info. - total_samples) / - block->data.stream_info.sample_rate + 0.5; - } - } while (FLAC__metadata_iterator_next(it)); - FLAC__metadata_iterator_delete(it); - - if (!tag_is_defined(ret)) { - tag_free(ret); - ret = NULL; - } - -out: - FLAC__metadata_chain_delete(chain); - return ret; -} - -static void -oggflac_decode(struct decoder *decoder, struct input_stream *input_stream) -{ - if (ogg_stream_type_detect(input_stream) != FLAC) - return; - - /* rewind the stream, because ogg_stream_type_detect() has - moved it */ - input_stream_seek(input_stream, 0, SEEK_SET); - - flac_decode_internal(decoder, input_stream, true); -} - -static const char *const oggflac_suffixes[] = { "ogg", "oga", NULL }; -static const char *const oggflac_mime_types[] = { - "application/ogg", - "application/x-ogg", - "audio/ogg", - "audio/x-flac+ogg", - "audio/x-ogg", - NULL -}; - -#endif /* FLAC_API_VERSION_CURRENT >= 7 */ - -const struct decoder_plugin oggflac_decoder_plugin = { - .name = "oggflac", - .init = oggflac_init, -#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7 - .stream_decode = oggflac_decode, - .tag_dup = oggflac_tag_dup, - .suffixes = oggflac_suffixes, - .mime_types = oggflac_mime_types -#endif -}; - -#endif /* HAVE_OGGFLAC */ - -static const char *const flac_suffixes[] = { "flac", NULL }; -static const char *const flac_mime_types[] = { - "application/flac", - "application/x-flac", - "audio/flac", - "audio/x-flac", - NULL -}; - -const struct decoder_plugin flac_decoder_plugin = { - .name = "flac", - .stream_decode = flac_decode, -#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7 - .file_decode = flac_filedecode, -#endif /* FLAC_API_VERSION_CURRENT >= 7 */ - .tag_dup = flac_tag_dup, - .suffixes = flac_suffixes, - .mime_types = flac_mime_types, -#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7 - .container_scan = flac_cue_track, -#endif /* FLAC_API_VERSION_CURRENT >= 7 */ -}; diff --git a/src/decoder/fluidsynth_plugin.c b/src/decoder/fluidsynth_decoder_plugin.c index 99c874c09..b9a2d0d99 100644 --- a/src/decoder/fluidsynth_plugin.c +++ b/src/decoder/fluidsynth_decoder_plugin.c @@ -1,5 +1,5 @@ /* - * Copyright (C) 2003-2009 The Music Player Daemon Project + * Copyright (C) 2003-2010 The Music Player Daemon Project * http://www.musicpd.org * * This program is free software; you can redistribute it and/or modify @@ -26,9 +26,10 @@ * */ -#include "../decoder_api.h" -#include "../timer.h" -#include "../conf.h" +#include "config.h" +#include "decoder_api.h" +#include "timer.h" +#include "conf.h" #include <glib.h> @@ -87,7 +88,7 @@ fluidsynth_file_decode(struct decoder *decoder, const char *path_fs) { static const struct audio_format audio_format = { .sample_rate = 48000, - .bits = 16, + .format = SAMPLE_FORMAT_S16, .channels = 2, }; char setting_sample_rate[] = "synth.sample-rate"; @@ -203,7 +204,7 @@ fluidsynth_file_decode(struct decoder *decoder, const char *path_fs) break; cmd = decoder_data(decoder, NULL, buffer, sizeof(buffer), - 0, 0, NULL); + 0); } while (cmd == DECODE_COMMAND_NONE); /* clean up */ diff --git a/src/decoder/gme_decoder_plugin.c b/src/decoder/gme_decoder_plugin.c new file mode 100644 index 000000000..e14a52d32 --- /dev/null +++ b/src/decoder/gme_decoder_plugin.c @@ -0,0 +1,247 @@ +#include "config.h" +#include "../decoder_api.h" +#include "audio_check.h" +#include "uri.h" + +#include <glib.h> +#include <assert.h> +#include <errno.h> +#include <stdlib.h> +#include <string.h> + +#include <gme/gme.h> + +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "gme" + +#define SUBTUNE_PREFIX "tune_" + +enum { + GME_SAMPLE_RATE = 44100, + GME_CHANNELS = 2, + GME_BUFFER_FRAMES = 2048, + GME_BUFFER_SAMPLES = GME_BUFFER_FRAMES * GME_CHANNELS, +}; + +/** + * returns the file path stripped of any /tune_xxx.* subtune + * suffix + */ +static char * +get_container_name(const char *path_fs) +{ + const char *subtune_suffix = uri_get_suffix(path_fs); + char *path_container = g_strdup(path_fs); + char *pat = g_strconcat("*/" SUBTUNE_PREFIX "???.", subtune_suffix, NULL); + GPatternSpec *path_with_subtune = g_pattern_spec_new(pat); + g_free(pat); + if (!g_pattern_match(path_with_subtune, + strlen(path_container), path_container, NULL)) { + g_pattern_spec_free(path_with_subtune); + return path_container; + } + + char *ptr = g_strrstr(path_container, "/" SUBTUNE_PREFIX); + if (ptr != NULL) + *ptr='\0'; + + g_pattern_spec_free(path_with_subtune); + return path_container; +} + +/** + * returns tune number from file.nsf/tune_xxx.* style path or 0 if no subtune + * is appended. + */ +static int +get_song_num(const char *path_fs) +{ + const char *subtune_suffix = uri_get_suffix(path_fs); + char *pat = g_strconcat("*/" SUBTUNE_PREFIX "???.", subtune_suffix, NULL); + GPatternSpec *path_with_subtune = g_pattern_spec_new(pat); + g_free(pat); + + if (g_pattern_match(path_with_subtune, + strlen(path_fs), path_fs, NULL)) { + char *sub = g_strrstr(path_fs, "/" SUBTUNE_PREFIX); + g_pattern_spec_free(path_with_subtune); + if(!sub) + return 0; + + sub += strlen("/" SUBTUNE_PREFIX); + int song_num = strtol(sub, NULL, 10); + + return song_num - 1; + } else { + g_pattern_spec_free(path_with_subtune); + return 0; + } +} + +static char * +gme_container_scan(const char *path_fs, const unsigned int tnum) +{ + Music_Emu *emu; + const char* gme_err; + unsigned int num_songs; + + gme_err = gme_open_file(path_fs, &emu, GME_SAMPLE_RATE); + if (gme_err != NULL) { + g_warning("%s", gme_err); + return NULL; + } + + num_songs = gme_track_count(emu); + /* if it only contains a single tune, don't treat as container */ + if (num_songs < 2) + return NULL; + + const char *subtune_suffix = uri_get_suffix(path_fs); + if (tnum <= num_songs){ + char *subtune = g_strdup_printf( + SUBTUNE_PREFIX "%03u.%s", tnum, subtune_suffix); + return subtune; + } else + return NULL; +} + +static void +gme_file_decode(struct decoder *decoder, const char *path_fs) +{ + float song_len; + Music_Emu *emu; + gme_info_t *ti; + struct audio_format audio_format; + enum decoder_command cmd; + short buf[GME_BUFFER_SAMPLES]; + const char* gme_err; + char *path_container = get_container_name(path_fs); + int song_num = get_song_num(path_fs); + + gme_err = gme_open_file(path_container, &emu, GME_SAMPLE_RATE); + g_free(path_container); + if (gme_err != NULL) { + g_warning("%s", gme_err); + return; + } + + if((gme_err = gme_track_info(emu, &ti, song_num)) != NULL){ + g_warning("%s", gme_err); + gme_delete(emu); + return; + } + + if(ti->length > 0) + song_len = ti->length / 1000.0; + else song_len = -1; + + /* initialize the MPD decoder */ + + GError *error = NULL; + if (!audio_format_init_checked(&audio_format, GME_SAMPLE_RATE, + SAMPLE_FORMAT_S16, GME_CHANNELS, + &error)) { + g_warning("%s", error->message); + g_error_free(error); + gme_free_info(ti); + gme_delete(emu); + return; + } + + decoder_initialized(decoder, &audio_format, true, song_len); + + if((gme_err = gme_start_track(emu, song_num)) != NULL) + g_warning("%s", gme_err); + + if(ti->length > 0) + gme_set_fade(emu, ti->length); + + /* play */ + do { + gme_err = gme_play(emu, GME_BUFFER_SAMPLES, buf); + if (gme_err != NULL) { + g_warning("%s", gme_err); + return; + } + cmd = decoder_data(decoder, NULL, buf, sizeof(buf), 0); + + if(cmd == DECODE_COMMAND_SEEK) { + float where = decoder_seek_where(decoder); + if((gme_err = gme_seek(emu, (int)where*1000)) != NULL) + g_warning("%s", gme_err); + decoder_command_finished(decoder); + } + + if(gme_track_ended(emu)) + break; + } while(cmd != DECODE_COMMAND_STOP); + + gme_free_info(ti); + gme_delete(emu); +} + +static struct tag * +gme_tag_dup(const char *path_fs) +{ + Music_Emu *emu; + gme_info_t *ti; + const char* gme_err; + char *path_container=get_container_name(path_fs); + int song_num; + song_num=get_song_num(path_fs); + + gme_err = gme_open_file(path_container, &emu, GME_SAMPLE_RATE); + g_free(path_container); + if (gme_err != NULL) { + g_warning("%s", gme_err); + return NULL; + } + if((gme_err = gme_track_info(emu, &ti, song_num)) != NULL){ + g_warning("%s", gme_err); + gme_delete(emu); + return NULL; + } + + struct tag *tag = tag_new(); + if(ti != NULL){ + if(ti->length > 0) + tag->time = ti->length / 1000; + if(ti->song != NULL){ + if(gme_track_count(emu) > 1){ + /* start numbering subtunes from 1 */ + char *tag_title=g_strdup_printf("%s (%d/%d)", + ti->song, song_num+1, gme_track_count(emu)); + tag_add_item(tag, TAG_TITLE, tag_title); + g_free(tag_title); + }else + tag_add_item(tag, TAG_TITLE, ti->song); + } + if(ti->author != NULL) + tag_add_item(tag, TAG_ARTIST, ti->author); + if(ti->game != NULL) + tag_add_item(tag, TAG_ALBUM, ti->game); + if(ti->comment != NULL) + tag_add_item(tag, TAG_COMMENT, ti->comment); + if(ti->copyright != NULL) + tag_add_item(tag, TAG_DATE, ti->copyright); + } + + gme_free_info(ti); + gme_delete(emu); + return tag; +} + +static const char *const gme_suffixes[] = { + "ay", "gbs", "gym", "hes", "kss", "nsf", + "nsfe", "sap", "spc", "vgm", "vgz", + NULL +}; + +extern const struct decoder_plugin gme_decoder_plugin; +const struct decoder_plugin gme_decoder_plugin = { + .name = "gme", + .file_decode = gme_file_decode, + .tag_dup = gme_tag_dup, + .suffixes = gme_suffixes, + .container_scan = gme_container_scan, +}; diff --git a/src/decoder/mad_plugin.c b/src/decoder/mad_decoder_plugin.c index 9b3259485..2c2906c5c 100644 --- a/src/decoder/mad_plugin.c +++ b/src/decoder/mad_decoder_plugin.c @@ -1,5 +1,5 @@ /* - * Copyright (C) 2003-2009 The Music Player Daemon Project + * Copyright (C) 2003-2010 The Music Player Daemon Project * http://www.musicpd.org * * This program is free software; you can redistribute it and/or modify @@ -17,10 +17,12 @@ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ -#include "../decoder_api.h" -#include "../conf.h" #include "config.h" +#include "decoder_api.h" +#include "conf.h" #include "tag_id3.h" +#include "tag_rva2.h" +#include "audio_check.h" #include <assert.h> #include <unistd.h> @@ -125,6 +127,7 @@ struct mp3_data { unsigned int drop_end_frames; unsigned int drop_start_samples; unsigned int drop_end_samples; + bool found_replay_gain; bool found_xing; bool found_first_frame; bool decoded_first_frame; @@ -148,6 +151,7 @@ mp3_data_init(struct mp3_data *data, struct decoder *decoder, data->drop_end_frames = 0; data->drop_start_samples = 0; data->drop_end_samples = 0; + data->found_replay_gain = false; data->found_xing = false; data->found_first_frame = false; data->decoded_first_frame = false; @@ -164,7 +168,7 @@ mp3_data_init(struct mp3_data *data, struct decoder *decoder, static bool mp3_seek(struct mp3_data *data, long offset) { - if (!input_stream_seek(data->input_stream, offset, SEEK_SET)) + if (!input_stream_seek(data->input_stream, offset, SEEK_SET, NULL)) return false; mad_stream_buffer(&data->stream, data->input_buffer, 0); @@ -208,105 +212,17 @@ mp3_fill_buffer(struct mp3_data *data) } #ifdef HAVE_ID3TAG -/* Parse mp3 RVA2 frame. Shamelessly stolen from madplay. */ static bool -parse_rva2(struct id3_tag *tag, struct replay_gain_info *replay_gain_info) -{ - struct id3_frame const * frame; - - id3_latin1_t const *id; - id3_byte_t const *data; - id3_length_t length; - - enum { - CHANNEL_OTHER = 0x00, - CHANNEL_MASTER_VOLUME = 0x01, - CHANNEL_FRONT_RIGHT = 0x02, - CHANNEL_FRONT_LEFT = 0x03, - CHANNEL_BACK_RIGHT = 0x04, - CHANNEL_BACK_LEFT = 0x05, - CHANNEL_FRONT_CENTRE = 0x06, - CHANNEL_BACK_CENTRE = 0x07, - CHANNEL_SUBWOOFER = 0x08 - }; - - /* relative volume adjustment information */ - - frame = id3_tag_findframe(tag, "RVA2", 0); - if (frame == NULL) - return false; - - id = id3_field_getlatin1(id3_frame_field(frame, 0)); - data = id3_field_getbinarydata(id3_frame_field(frame, 1), - &length); - - if (id == NULL || data == NULL) - return false; - - /* - * "The 'identification' string is used to identify the - * situation and/or device where this adjustment should apply. - * The following is then repeated for every channel - * - * Type of channel $xx - * Volume adjustment $xx xx - * Bits representing peak $xx - * Peak volume $xx (xx ...)" - */ - - while (length >= 4) { - unsigned int peak_bytes; - - peak_bytes = (data[3] + 7) / 8; - if (4 + peak_bytes > length) - break; - - if (data[0] == CHANNEL_MASTER_VOLUME) { - signed int voladj_fixed; - double voladj_float; - - /* - * "The volume adjustment is encoded as a fixed - * point decibel value, 16 bit signed integer - * representing (adjustment*512), giving +/- 64 - * dB with a precision of 0.001953125 dB." - */ - - voladj_fixed = (data[1] << 8) | (data[2] << 0); - voladj_fixed |= -(voladj_fixed & 0x8000); - - voladj_float = (double) voladj_fixed / 512; - - replay_gain_info->tuples[REPLAY_GAIN_TRACK].gain = voladj_float; - replay_gain_info->tuples[REPLAY_GAIN_ALBUM].gain = voladj_float; - - g_debug("parseRVA2: Relative Volume " - "%+.1f dB adjustment (%s)\n", - voladj_float, id); - - return true; - } - - data += 4 + peak_bytes; - length -= 4 + peak_bytes; - } - - return false; -} -#endif - -#ifdef HAVE_ID3TAG -static struct replay_gain_info * -parse_id3_replay_gain_info(struct id3_tag *tag) +parse_id3_replay_gain_info(struct replay_gain_info *replay_gain_info, + struct id3_tag *tag) { int i; char *key; char *value; struct id3_frame *frame; bool found = false; - struct replay_gain_info *replay_gain_info; - replay_gain_info = replay_gain_info_new(); + replay_gain_info_init(replay_gain_info); for (i = 0; (frame = id3_tag_findframe(tag, "TXXX", i)); i++) { if (frame->nfields < 3) @@ -337,21 +253,55 @@ parse_id3_replay_gain_info(struct id3_tag *tag) free(value); } - if (!found) { + return found || /* fall back on RVA2 if no replaygain tags found */ - found = parse_rva2(tag, replay_gain_info); + tag_rva2_parse(tag, replay_gain_info); +} +#endif + +#ifdef HAVE_ID3TAG +static bool +parse_id3_mixramp(char **mixramp_start, char **mixramp_end, + struct id3_tag *tag) +{ + int i; + char *key; + char *value; + struct id3_frame *frame; + bool found = false; + + *mixramp_start = NULL; + *mixramp_end = NULL; + + for (i = 0; (frame = id3_tag_findframe(tag, "TXXX", i)); i++) { + if (frame->nfields < 3) + continue; + + key = (char *) + id3_ucs4_latin1duplicate(id3_field_getstring + (&frame->fields[1])); + value = (char *) + id3_ucs4_latin1duplicate(id3_field_getstring + (&frame->fields[2])); + + if (g_ascii_strcasecmp(key, "mixramp_start") == 0) { + *mixramp_start = g_strdup(value); + found = true; + } else if (g_ascii_strcasecmp(key, "mixramp_end") == 0) { + *mixramp_end = g_strdup(value); + found = true; + } + + free(key); + free(value); } - if (found) - return replay_gain_info; - replay_gain_info_free(replay_gain_info); - return NULL; + return found; } #endif static void mp3_parse_id3(struct mp3_data *data, size_t tagsize, - struct tag **mpd_tag, - struct replay_gain_info **replay_gain_info_r) + struct tag **mpd_tag) { #ifdef HAVE_ID3TAG struct id3_tag *id3_tag = NULL; @@ -404,13 +354,20 @@ static void mp3_parse_id3(struct mp3_data *data, size_t tagsize, } } - if (replay_gain_info_r) { - struct replay_gain_info *tmp_rgi = - parse_id3_replay_gain_info(id3_tag); - if (tmp_rgi != NULL) { - if (*replay_gain_info_r) - replay_gain_info_free(*replay_gain_info_r); - *replay_gain_info_r = tmp_rgi; + if (data->decoder != NULL) { + struct replay_gain_info rgi; + char *mixramp_start; + char *mixramp_end; + float replay_gain_db = 0; + + if (parse_id3_replay_gain_info(&rgi, id3_tag)) { + replay_gain_db = decoder_replay_gain(data->decoder, &rgi); + data->found_replay_gain = true; + } + if (parse_id3_mixramp(&mixramp_start, &mixramp_end, id3_tag)) { + g_debug("setting mixramp_tags"); + decoder_mixramp(data->decoder, replay_gain_db, + mixramp_start, mixramp_end); } } @@ -419,7 +376,6 @@ static void mp3_parse_id3(struct mp3_data *data, size_t tagsize, g_free(allocated); #else /* !HAVE_ID3TAG */ (void)mpd_tag; - (void)replay_gain_info_r; /* This code is enabled when libid3tag is disabled. Instead of parsing the ID3 frame, it just skips it. */ @@ -467,8 +423,7 @@ id3_tag_query(const void *p0, size_t length) #endif /* !HAVE_ID3TAG */ static enum mp3_action -decode_next_frame_header(struct mp3_data *data, G_GNUC_UNUSED struct tag **tag, - G_GNUC_UNUSED struct replay_gain_info **replay_gain_info_r) +decode_next_frame_header(struct mp3_data *data, G_GNUC_UNUSED struct tag **tag) { enum mad_layer layer; @@ -490,7 +445,7 @@ decode_next_frame_header(struct mp3_data *data, G_GNUC_UNUSED struct tag **tag, if (tagsize > 0) { if (tag && !(*tag)) { mp3_parse_id3(data, (size_t)tagsize, - tag, replay_gain_info_r); + tag); } else { mad_stream_skip(&(data->stream), tagsize); @@ -592,14 +547,14 @@ enum { XING_SCALE = 0x00000008L }; -struct version { +struct lame_version { unsigned major; unsigned minor; }; struct lame { char encoder[10]; /* 9 byte encoder name/version ("LAME3.97b") */ - struct version version; /* struct containing just the version */ + struct lame_version version; /* struct containing just the version */ float peak; /* replaygain peak */ float track_gain; /* replaygain track gain */ float album_gain; /* replaygain album gain */ @@ -798,10 +753,10 @@ mp3_frame_duration(const struct mad_frame *frame) MAD_UNITS_MILLISECONDS) / 1000.0; } -static off_t +static goffset mp3_this_frame_offset(const struct mp3_data *data) { - off_t offset = data->input_stream->offset; + goffset offset = data->input_stream->offset; if (data->stream.this_frame != NULL) offset -= data->stream.bufend - data->stream.this_frame; @@ -811,7 +766,7 @@ mp3_this_frame_offset(const struct mp3_data *data) return offset; } -static off_t +static goffset mp3_rest_including_this_frame(const struct mp3_data *data) { return data->input_stream->size - mp3_this_frame_offset(data); @@ -823,7 +778,7 @@ mp3_rest_including_this_frame(const struct mp3_data *data) static void mp3_filesize_to_song_length(struct mp3_data *data) { - off_t rest = mp3_rest_including_this_frame(data); + goffset rest = mp3_rest_including_this_frame(data); if (rest > 0) { float frame_duration = mp3_frame_duration(&data->frame); @@ -838,8 +793,7 @@ mp3_filesize_to_song_length(struct mp3_data *data) } static bool -mp3_decode_first_frame(struct mp3_data *data, struct tag **tag, - struct replay_gain_info **replay_gain_info_r) +mp3_decode_first_frame(struct mp3_data *data, struct tag **tag) { struct xing xing; struct lame lame; @@ -853,8 +807,7 @@ mp3_decode_first_frame(struct mp3_data *data, struct tag **tag, while (true) { do { - ret = decode_next_frame_header(data, tag, - replay_gain_info_r); + ret = decode_next_frame_header(data, tag); } while (ret == DECODE_CONT); if (ret == DECODE_BREAK) return false; @@ -897,14 +850,17 @@ mp3_decode_first_frame(struct mp3_data *data, struct tag **tag, /* Album gain isn't currently used. See comment in * parse_lame() for details. -- jat */ - if (replay_gain_info_r && !*replay_gain_info_r && + if (data->decoder != NULL && + !data->found_replay_gain && lame.track_gain) { - *replay_gain_info_r = replay_gain_info_new(); - (*replay_gain_info_r)->tuples[REPLAY_GAIN_TRACK].gain = lame.track_gain; - (*replay_gain_info_r)->tuples[REPLAY_GAIN_TRACK].peak = lame.peak; + struct replay_gain_info rgi; + replay_gain_info_init(&rgi); + rgi.tuples[REPLAY_GAIN_TRACK].gain = lame.track_gain; + rgi.tuples[REPLAY_GAIN_TRACK].peak = lame.peak; + decoder_replay_gain(data->decoder, &rgi); } } - } + } if (!data->max_frames) return false; @@ -932,33 +888,29 @@ static void mp3_data_finish(struct mp3_data *data) } /* this is primarily used for getting total time for tags */ -static int mp3_total_file_time(const char *file) +static int +mad_decoder_total_file_time(struct input_stream *is) { - struct input_stream input_stream; struct mp3_data data; int ret; - if (!input_stream_open(&input_stream, file)) - return -1; - mp3_data_init(&data, NULL, &input_stream); - if (!mp3_decode_first_frame(&data, NULL, NULL)) + mp3_data_init(&data, NULL, is); + if (!mp3_decode_first_frame(&data, NULL)) ret = -1; else ret = data.total_time + 0.5; mp3_data_finish(&data); - input_stream_close(&input_stream); return ret; } static bool mp3_open(struct input_stream *is, struct mp3_data *data, - struct decoder *decoder, struct tag **tag, - struct replay_gain_info **replay_gain_info_r) + struct decoder *decoder, struct tag **tag) { mp3_data_init(data, decoder, is); *tag = NULL; - if (!mp3_decode_first_frame(data, tag, replay_gain_info_r)) { + if (!mp3_decode_first_frame(data, tag)) { mp3_data_finish(data); if (tag && *tag) tag_free(*tag); @@ -1017,8 +969,7 @@ mp3_update_timer_next_frame(struct mp3_data *data) * Sends the synthesized current frame via decoder_data(). */ static enum decoder_command -mp3_send_pcm(struct mp3_data *data, unsigned i, unsigned pcm_length, - struct replay_gain_info *replay_gain_info) +mp3_send_pcm(struct mp3_data *data, unsigned i, unsigned pcm_length) { unsigned max_samples; @@ -1043,9 +994,7 @@ mp3_send_pcm(struct mp3_data *data, unsigned i, unsigned pcm_length, cmd = decoder_data(data->decoder, data->input_stream, data->output_buffer, sizeof(data->output_buffer[0]) * num_samples, - data->elapsed_time, - data->bit_rate / 1000, - replay_gain_info); + data->bit_rate / 1000); if (cmd != DECODE_COMMAND_NONE) return cmd; } @@ -1057,8 +1006,7 @@ mp3_send_pcm(struct mp3_data *data, unsigned i, unsigned pcm_length, * Synthesize the current frame and send it via decoder_data(). */ static enum decoder_command -mp3_synth_and_send(struct mp3_data *data, - struct replay_gain_info *replay_gain_info) +mp3_synth_and_send(struct mp3_data *data) { unsigned i, pcm_length; enum decoder_command cmd; @@ -1099,7 +1047,7 @@ mp3_synth_and_send(struct mp3_data *data, pcm_length -= data->drop_end_samples; } - cmd = mp3_send_pcm(data, i, pcm_length, replay_gain_info); + cmd = mp3_send_pcm(data, i, pcm_length); if (cmd != DECODE_COMMAND_NONE) return cmd; @@ -1113,7 +1061,7 @@ mp3_synth_and_send(struct mp3_data *data, } static bool -mp3_read(struct mp3_data *data, struct replay_gain_info **replay_gain_info_r) +mp3_read(struct mp3_data *data) { struct decoder *decoder = data->decoder; enum mp3_action ret; @@ -1130,9 +1078,7 @@ mp3_read(struct mp3_data *data, struct replay_gain_info **replay_gain_info_r) data->mute_frame = MUTEFRAME_NONE; break; case MUTEFRAME_NONE: - cmd = mp3_synth_and_send(data, - replay_gain_info_r != NULL - ? *replay_gain_info_r : NULL); + cmd = mp3_synth_and_send(data); if (cmd == DECODE_COMMAND_SEEK) { unsigned long j; @@ -1161,8 +1107,7 @@ mp3_read(struct mp3_data *data, struct replay_gain_info **replay_gain_info_r) do { struct tag *tag = NULL; - ret = decode_next_frame_header(data, &tag, - replay_gain_info_r); + ret = decode_next_frame_header(data, &tag); if (tag != NULL) { decoder_tag(decoder, data->input_stream, tag); @@ -1189,29 +1134,34 @@ mp3_read(struct mp3_data *data, struct replay_gain_info **replay_gain_info_r) return ret != DECODE_BREAK; } -static void mp3_audio_format(struct mp3_data *data, struct audio_format *af) -{ - af->bits = 24; - af->sample_rate = (data->frame).header.samplerate; - af->channels = MAD_NCHANNELS(&(data->frame).header); -} - static void mp3_decode(struct decoder *decoder, struct input_stream *input_stream) { struct mp3_data data; + GError *error = NULL; struct tag *tag = NULL; - struct replay_gain_info *replay_gain_info = NULL; struct audio_format audio_format; - if (!mp3_open(input_stream, &data, decoder, &tag, &replay_gain_info)) { + if (!mp3_open(input_stream, &data, decoder, &tag)) { if (decoder_get_command(decoder) == DECODE_COMMAND_NONE) g_warning ("Input does not appear to be a mp3 bit stream.\n"); return; } - mp3_audio_format(&data, &audio_format); + if (!audio_format_init_checked(&audio_format, + data.frame.header.samplerate, + SAMPLE_FORMAT_S24_P32, + MAD_NCHANNELS(&data.frame.header), + &error)) { + g_warning("%s", error->message); + g_error_free(error); + + if (tag != NULL) + tag_free(tag); + mp3_data_finish(&data); + return; + } decoder_initialized(decoder, &audio_format, data.input_stream->seekable, data.total_time); @@ -1221,24 +1171,20 @@ mp3_decode(struct decoder *decoder, struct input_stream *input_stream) tag_free(tag); } - while (mp3_read(&data, &replay_gain_info)) ; - - if (replay_gain_info) - replay_gain_info_free(replay_gain_info); + while (mp3_read(&data)) ; mp3_data_finish(&data); } -static struct tag *mp3_tag_dup(const char *file) +static struct tag * +mad_decoder_stream_tag(struct input_stream *is) { struct tag *tag; int total_time; - total_time = mp3_total_file_time(file); - if (total_time < 0) { - g_debug("Failed to get total song time from: %s", file); + total_time = mad_decoder_total_file_time(is); + if (total_time < 0) return NULL; - } tag = tag_new(); tag->time = total_time; @@ -1252,7 +1198,7 @@ const struct decoder_plugin mad_decoder_plugin = { .name = "mad", .init = mp3_plugin_init, .stream_decode = mp3_decode, - .tag_dup = mp3_tag_dup, + .stream_tag = mad_decoder_stream_tag, .suffixes = mp3_suffixes, .mime_types = mp3_mime_types }; diff --git a/src/decoder/mikmod_plugin.c b/src/decoder/mikmod_decoder_plugin.c index f60dcbc61..91478e86f 100644 --- a/src/decoder/mikmod_plugin.c +++ b/src/decoder/mikmod_decoder_plugin.c @@ -1,5 +1,5 @@ /* - * Copyright (C) 2003-2009 The Music Player Daemon Project + * Copyright (C) 2003-2010 The Music Player Daemon Project * http://www.musicpd.org * * This program is free software; you can redistribute it and/or modify @@ -17,10 +17,13 @@ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ -#include "../decoder_api.h" +#include "config.h" +#include "decoder_api.h" +#include "mpd_error.h" #include <glib.h> #include <mikmod.h> +#include <assert.h> #undef G_LOG_DOMAIN #define G_LOG_DOMAIN "mikmod" @@ -29,30 +32,34 @@ #define MIKMOD_FRAME_SIZE 4096 -static BOOL mod_mpd_Init(void) +static BOOL +mikmod_mpd_init(void) { return VC_Init(); } -static void mod_mpd_Exit(void) +static void +mikmod_mpd_exit(void) { VC_Exit(); } -static void mod_mpd_Update(void) +static void +mikmod_mpd_update(void) { } -static BOOL mod_mpd_IsThere(void) +static BOOL +mikmod_mpd_is_present(void) { - return 1; + return true; } -static char drv_name[] = "MPD"; -static char drv_version[] = "MPD Output Driver v0.1"; +static char drv_name[] = PACKAGE_NAME; +static char drv_version[] = VERSION; #if (LIBMIKMOD_VERSION > 0x030106) -static char drv_alias[] = "mpd"; +static char drv_alias[] = PACKAGE; #endif static MDRIVER drv_mpd = { @@ -68,18 +75,18 @@ static MDRIVER drv_mpd = { #endif NULL, /* CommandLine */ #endif - mod_mpd_IsThere, + mikmod_mpd_is_present, VC_SampleLoad, VC_SampleUnload, VC_SampleSpace, VC_SampleLength, - mod_mpd_Init, - mod_mpd_Exit, + mikmod_mpd_init, + mikmod_mpd_exit, NULL, VC_SetNumVoices, VC_PlayStart, VC_PlayStop, - mod_mpd_Update, + mikmod_mpd_update, NULL, VC_VoiceSetVolume, VC_VoiceGetVolume, @@ -94,11 +101,19 @@ static MDRIVER drv_mpd = { VC_VoiceRealVolume }; +static unsigned mikmod_sample_rate; + static bool -mod_initMikMod(G_GNUC_UNUSED const struct config_param *param) +mikmod_decoder_init(const struct config_param *param) { static char params[] = ""; + mikmod_sample_rate = config_get_block_unsigned(param, "sample_rate", + 44100); + if (!audio_valid_sample_rate(mikmod_sample_rate)) + MPD_ERROR("Invalid sample rate in line %d: %u", + param->line, mikmod_sample_rate); + md_device = 0; md_reverb = 0; @@ -106,7 +121,7 @@ mod_initMikMod(G_GNUC_UNUSED const struct config_param *param) MikMod_RegisterAllLoaders(); md_pansep = 64; - md_mixfreq = 44100; + md_mixfreq = mikmod_sample_rate; md_mode = (DMODE_SOFT_MUSIC | DMODE_INTERP | DMODE_STEREO | DMODE_16BITS); @@ -119,117 +134,80 @@ mod_initMikMod(G_GNUC_UNUSED const struct config_param *param) return true; } -static void mod_finishMikMod(void) +static void +mikmod_decoder_finish(void) { MikMod_Exit(); } -typedef struct _mod_Data { - MODULE *moduleHandle; - SBYTE audio_buffer[MIKMOD_FRAME_SIZE]; -} mod_Data; - -static mod_Data *mod_open(const char *path) -{ - char *path2; - MODULE *moduleHandle; - mod_Data *data; - - path2 = g_strdup(path); - moduleHandle = Player_Load(path2, 128, 0); - g_free(path2); - - if (moduleHandle == NULL) - return NULL; - - /* Prevent module from looping forever */ - moduleHandle->loop = 0; - - data = g_new(mod_Data, 1); - data->moduleHandle = moduleHandle; - - Player_Start(data->moduleHandle); - - return data; -} - -static void mod_close(mod_Data * data) -{ - Player_Stop(); - Player_Free(data->moduleHandle); - g_free(data); -} - static void -mod_decode(struct decoder *decoder, const char *path) +mikmod_decoder_file_decode(struct decoder *decoder, const char *path_fs) { - mod_Data *data; + char *path2; + MODULE *handle; struct audio_format audio_format; - float total_time = 0.0; int ret; - float secPerByte; + SBYTE buffer[MIKMOD_FRAME_SIZE]; enum decoder_command cmd = DECODE_COMMAND_NONE; - if (!(data = mod_open(path))) { - g_warning("failed to open mod: %s\n", path); + path2 = g_strdup(path_fs); + handle = Player_Load(path2, 128, 0); + g_free(path2); + + if (handle == NULL) { + g_warning("failed to open mod: %s", path_fs); return; } - audio_format.bits = 16; - audio_format.sample_rate = 44100; - audio_format.channels = 2; + /* Prevent module from looping forever */ + handle->loop = 0; - secPerByte = - 1.0 / ((audio_format.bits * audio_format.channels / 8.0) * - (float)audio_format.sample_rate); + audio_format_init(&audio_format, mikmod_sample_rate, SAMPLE_FORMAT_S16, 2); + assert(audio_format_valid(&audio_format)); decoder_initialized(decoder, &audio_format, false, 0); + Player_Start(handle); while (cmd == DECODE_COMMAND_NONE && Player_Active()) { - ret = VC_WriteBytes(data->audio_buffer, MIKMOD_FRAME_SIZE); - total_time += ret * secPerByte; - cmd = decoder_data(decoder, NULL, - data->audio_buffer, ret, - total_time, 0, NULL); + ret = VC_WriteBytes(buffer, sizeof(buffer)); + cmd = decoder_data(decoder, NULL, buffer, ret, 0); } - mod_close(data); + Player_Stop(); + Player_Free(handle); } -static struct tag *modTagDup(const char *file) +static struct tag * +mikmod_decoder_tag_dup(const char *path_fs) { - char *path2; - struct tag *ret = NULL; - MODULE *moduleHandle; - char *title; + char *path2 = g_strdup(path_fs); + MODULE *handle = Player_Load(path2, 128, 0); - path2 = g_strdup(file); - moduleHandle = Player_Load(path2, 128, 0); - g_free(path2); - - if (moduleHandle == NULL) { - g_debug("Failed to open file: %s", file); + if (handle == NULL) { + g_free(path2); + g_debug("Failed to open file: %s", path_fs); return NULL; } - Player_Free(moduleHandle); - ret = tag_new(); + Player_Free(handle); - ret->time = 0; + struct tag *tag = tag_new(); - path2 = g_strdup(file); - title = Player_LoadTitle(path2); + tag->time = 0; + + char *title = Player_LoadTitle(path2); g_free(path2); - if (title) { - tag_add_item(ret, TAG_ITEM_TITLE, title); + + if (title != NULL) { + tag_add_item(tag, TAG_TITLE, title); free(title); } - return ret; + return tag; } -static const char *const modSuffixes[] = { +static const char *const mikmod_decoder_suffixes[] = { "amf", "dsm", "far", @@ -250,9 +228,9 @@ static const char *const modSuffixes[] = { const struct decoder_plugin mikmod_decoder_plugin = { .name = "mikmod", - .init = mod_initMikMod, - .finish = mod_finishMikMod, - .file_decode = mod_decode, - .tag_dup = modTagDup, - .suffixes = modSuffixes, + .init = mikmod_decoder_init, + .finish = mikmod_decoder_finish, + .file_decode = mikmod_decoder_file_decode, + .tag_dup = mikmod_decoder_tag_dup, + .suffixes = mikmod_decoder_suffixes, }; diff --git a/src/decoder/modplug_plugin.c b/src/decoder/modplug_decoder_plugin.c index f636f2fa6..037c2fd74 100644 --- a/src/decoder/modplug_plugin.c +++ b/src/decoder/modplug_decoder_plugin.c @@ -1,5 +1,5 @@ /* - * Copyright (C) 2003-2009 The Music Player Daemon Project + * Copyright (C) 2003-2010 The Music Player Daemon Project * http://www.musicpd.org * * This program is free software; you can redistribute it and/or modify @@ -17,10 +17,12 @@ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ -#include "../decoder_api.h" +#include "config.h" +#include "decoder_api.h" #include <glib.h> #include <modplug.h> +#include <assert.h> #undef G_LOG_DOMAIN #define G_LOG_DOMAIN "modplug" @@ -92,10 +94,8 @@ mod_decode(struct decoder *decoder, struct input_stream *is) ModPlug_Settings settings; GByteArray *bdatas; struct audio_format audio_format; - float total_time = 0.0; - int ret, current; + int ret; char audio_buffer[MODPLUG_FRAME_SIZE]; - float sec_perbyte; enum decoder_command cmd = DECODE_COMMAND_NONE; bdatas = mod_loadfile(decoder, is); @@ -121,37 +121,26 @@ mod_decode(struct decoder *decoder, struct input_stream *is) return; } - audio_format.bits = 16; - audio_format.sample_rate = 44100; - audio_format.channels = 2; - - sec_perbyte = - 1.0 / ((audio_format.bits * audio_format.channels / 8.0) * - (float)audio_format.sample_rate); - - total_time = ModPlug_GetLength(f) / 1000; + audio_format_init(&audio_format, 44100, SAMPLE_FORMAT_S16, 2); + assert(audio_format_valid(&audio_format)); decoder_initialized(decoder, &audio_format, - is->seekable, total_time); - - total_time = 0; + is->seekable, ModPlug_GetLength(f) / 1000.0); do { ret = ModPlug_Read(f, audio_buffer, MODPLUG_FRAME_SIZE); - - if (ret == 0) { + if (ret <= 0) break; - } - total_time += ret * sec_perbyte; cmd = decoder_data(decoder, NULL, audio_buffer, ret, - total_time, 0, NULL); + 0); if (cmd == DECODE_COMMAND_SEEK) { - total_time = decoder_seek_where(decoder); - current = total_time * 1000; - ModPlug_Seek(f, current); + float where = decoder_seek_where(decoder); + + ModPlug_Seek(f, (int)(where * 1000.0)); + decoder_command_finished(decoder); } @@ -160,43 +149,33 @@ mod_decode(struct decoder *decoder, struct input_stream *is) ModPlug_Unload(f); } -static struct tag *mod_tagdup(const char *file) +static struct tag * +modplug_stream_tag(struct input_stream *is) { ModPlugFile *f; struct tag *ret = NULL; GByteArray *bdatas; char *title; - struct input_stream is; - if (!input_stream_open(&is, file)) { - g_warning("cant open file %s\n", file); + bdatas = mod_loadfile(NULL, is); + if (!bdatas) return NULL; - } - - bdatas = mod_loadfile(NULL, &is); - if (!bdatas) { - g_warning("cant load file %s\n", file); - return NULL; - } f = ModPlug_Load(bdatas->data, bdatas->len); g_byte_array_free(bdatas, TRUE); - if (!f) { - g_warning("could not decode file %s\n", file); + if (f == NULL) return NULL; - } + ret = tag_new(); - ret->time = 0; + ret->time = ModPlug_GetLength(f) / 1000; title = g_strdup(ModPlug_GetName(f)); if (title) - tag_add_item(ret, TAG_ITEM_TITLE, title); + tag_add_item(ret, TAG_TITLE, title); g_free(title); ModPlug_Unload(f); - input_stream_close(&is); - return ret; } @@ -210,6 +189,6 @@ static const char *const mod_suffixes[] = { const struct decoder_plugin modplug_decoder_plugin = { .name = "modplug", .stream_decode = mod_decode, - .tag_dup = mod_tagdup, + .stream_tag = modplug_stream_tag, .suffixes = mod_suffixes, }; diff --git a/src/decoder/mp4ff_plugin.c b/src/decoder/mp4ff_decoder_plugin.c index 4d4d47c6c..861b08194 100644 --- a/src/decoder/mp4ff_plugin.c +++ b/src/decoder/mp4ff_decoder_plugin.c @@ -1,5 +1,5 @@ /* - * Copyright (C) 2003-2009 The Music Player Daemon Project + * Copyright (C) 2003-2010 The Music Player Daemon Project * http://www.musicpd.org * * This program is free software; you can redistribute it and/or modify @@ -17,8 +17,9 @@ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ -#include "../decoder_api.h" #include "config.h" +#include "decoder_api.h" +#include "audio_check.h" #include "tag_table.h" #include <glib.h> @@ -35,7 +36,9 @@ /* all code here is either based on or copied from FAAD2's frontend code */ -struct mp4_context { +struct mp4ff_input_stream { + mp4ff_callback_t callback; + struct decoder *decoder; struct input_stream *input_stream; }; @@ -89,20 +92,38 @@ mp4_get_aac_track(mp4ff_t * infile, faacDecHandle decoder, static uint32_t mp4_read(void *user_data, void *buffer, uint32_t length) { - struct mp4_context *ctx = user_data; + struct mp4ff_input_stream *mis = user_data; - return decoder_read(ctx->decoder, ctx->input_stream, buffer, length); + return decoder_read(mis->decoder, mis->input_stream, buffer, length); } static uint32_t mp4_seek(void *user_data, uint64_t position) { - struct mp4_context *ctx = user_data; + struct mp4ff_input_stream *mis = user_data; - return input_stream_seek(ctx->input_stream, position, SEEK_SET) + return input_stream_seek(mis->input_stream, position, SEEK_SET, NULL) ? 0 : -1; } +static const mp4ff_callback_t mpd_mp4ff_callback = { + .read = mp4_read, + .seek = mp4_seek, +}; + +static mp4ff_t * +mp4ff_input_stream_open(struct mp4ff_input_stream *mis, + struct decoder *decoder, + struct input_stream *input_stream) +{ + mis->callback = mpd_mp4ff_callback; + mis->callback.user_data = mis; + mis->decoder = decoder; + mis->input_stream = input_stream; + + return mp4ff_open_read(&mis->callback); +} + static faacDecHandle mp4_faad_new(mp4ff_t *mp4fh, int *track_r, struct audio_format *audio_format) { @@ -111,6 +132,7 @@ mp4_faad_new(mp4ff_t *mp4fh, int *track_r, struct audio_format *audio_format) int track; uint32_t sample_rate; unsigned char channels; + GError *error = NULL; decoder = faacDecOpen(); @@ -131,37 +153,24 @@ mp4_faad_new(mp4ff_t *mp4fh, int *track_r, struct audio_format *audio_format) return NULL; } - *track_r = track; - *audio_format = (struct audio_format){ - .bits = 16, - .channels = channels, - .sample_rate = sample_rate, - }; - - if (!audio_format_valid(audio_format)) { - g_warning("Invalid audio format: %u:%u:%u\n", - audio_format->sample_rate, - audio_format->bits, - audio_format->channels); + if (!audio_format_init_checked(audio_format, sample_rate, + SAMPLE_FORMAT_S16, channels, + &error)) { + g_warning("%s", error->message); + g_error_free(error); faacDecClose(decoder); return NULL; } + *track_r = track; + return decoder; } static void mp4_decode(struct decoder *mpd_decoder, struct input_stream *input_stream) { - struct mp4_context ctx = { - .decoder = mpd_decoder, - .input_stream = input_stream, - }; - mp4ff_callback_t callback = { - .read = mp4_read, - .seek = mp4_seek, - .user_data = &ctx, - }; + struct mp4ff_input_stream mis; mp4ff_t *mp4fh; int32_t track; float file_time, total_time; @@ -187,7 +196,7 @@ mp4_decode(struct decoder *mpd_decoder, struct input_stream *input_stream) double seek_where = 0; enum decoder_command cmd = DECODE_COMMAND_NONE; - mp4fh = mp4ff_open_read(&callback); + mp4fh = mp4ff_input_stream_open(&mis, mpd_decoder, input_stream); if (!mp4fh) { g_warning("Input does not appear to be a mp4 stream.\n"); return; @@ -266,7 +275,7 @@ mp4_decode(struct decoder *mpd_decoder, struct input_stream *input_stream) dur -= offset; file_time += ((float)dur) / scale; - if (seeking && file_time > seek_where) + if (seeking && file_time >= seek_where) seek_position_found = true; if (seeking && seek_position_found) { @@ -332,7 +341,7 @@ mp4_decode(struct decoder *mpd_decoder, struct input_stream *input_stream) cmd = decoder_data(mpd_decoder, input_stream, sample_buffer, sample_buffer_length, - file_time, bit_rate, NULL); + bit_rate); } g_free(seek_table); @@ -341,9 +350,9 @@ mp4_decode(struct decoder *mpd_decoder, struct input_stream *input_stream) } static const char *const mp4ff_tag_names[TAG_NUM_OF_ITEM_TYPES] = { - [TAG_ITEM_ALBUM_ARTIST] = "album artist", - [TAG_ITEM_COMPOSER] = "writer", - [TAG_ITEM_PERFORMER] = "band", + [TAG_ALBUM_ARTIST] = "album artist", + [TAG_COMPOSER] = "writer", + [TAG_PERFORMER] = "band", }; static enum tag_type @@ -353,57 +362,41 @@ mp4ff_tag_name_parse(const char *name) if (type == TAG_NUM_OF_ITEM_TYPES) type = tag_name_parse_i(name); + if (g_ascii_strcasecmp(name, "albumartist") == 0 || + g_ascii_strcasecmp(name, "album_artist") == 0) + return TAG_ALBUM_ARTIST; + return type; } static struct tag * -mp4_tag_dup(const char *file) +mp4_stream_tag(struct input_stream *is) { - struct tag *ret = NULL; - struct input_stream input_stream; - struct mp4_context ctx = { - .decoder = NULL, - .input_stream = &input_stream, - }; - mp4ff_callback_t callback = { - .read = mp4_read, - .seek = mp4_seek, - .user_data = &ctx, - }; - mp4ff_t *mp4fh; + struct mp4ff_input_stream mis; int32_t track; int32_t file_time; int32_t scale; int i; - if (!input_stream_open(&input_stream, file)) { - g_warning("Failed to open file: %s", file); + mp4ff_t *mp4fh = mp4ff_input_stream_open(&mis, NULL, is); + if (mp4fh == NULL) return NULL; - } - - mp4fh = mp4ff_open_read(&callback); - if (!mp4fh) { - input_stream_close(&input_stream); - return NULL; - } track = mp4_get_aac_track(mp4fh, NULL, NULL, NULL); if (track < 0) { mp4ff_close(mp4fh); - input_stream_close(&input_stream); return NULL; } - ret = tag_new(); file_time = mp4ff_get_track_duration_use_offsets(mp4fh, track); scale = mp4ff_time_scale(mp4fh, track); if (scale < 0) { mp4ff_close(mp4fh); - input_stream_close(&input_stream); - tag_free(ret); return NULL; } - ret->time = ((float)file_time) / scale + 0.5; + + struct tag *tag = tag_new(); + tag->time = ((float)file_time) / scale + 0.5; for (i = 0; i < mp4ff_meta_get_num_items(mp4fh); i++) { char *item; @@ -413,16 +406,15 @@ mp4_tag_dup(const char *file) enum tag_type type = mp4ff_tag_name_parse(item); if (type != TAG_NUM_OF_ITEM_TYPES) - tag_add_item(ret, type, value); + tag_add_item(tag, type, value); free(item); free(value); } mp4ff_close(mp4fh); - input_stream_close(&input_stream); - return ret; + return tag; } static const char *const mp4_suffixes[] = { @@ -435,9 +427,9 @@ static const char *const mp4_suffixes[] = { static const char *const mp4_mime_types[] = { "audio/mp4", "audio/m4a", NULL }; const struct decoder_plugin mp4ff_decoder_plugin = { - .name = "mp4", + .name = "mp4ff", .stream_decode = mp4_decode, - .tag_dup = mp4_tag_dup, + .stream_tag = mp4_stream_tag, .suffixes = mp4_suffixes, .mime_types = mp4_mime_types, }; diff --git a/src/decoder/mpcdec_plugin.c b/src/decoder/mpcdec_decoder_plugin.c index 72a516f22..4df8dd218 100644 --- a/src/decoder/mpcdec_plugin.c +++ b/src/decoder/mpcdec_decoder_plugin.c @@ -1,5 +1,5 @@ /* - * Copyright (C) 2003-2009 The Music Player Daemon Project + * Copyright (C) 2003-2010 The Music Player Daemon Project * http://www.musicpd.org * * This program is free software; you can redistribute it and/or modify @@ -17,8 +17,9 @@ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ -#include "../decoder_api.h" #include "config.h" +#include "decoder_api.h" +#include "audio_check.h" #ifdef MPC_IS_OLD_API #include <mpcdec/mpcdec.h> @@ -28,6 +29,7 @@ #endif #include <glib.h> +#include <assert.h> #include <unistd.h> #undef G_LOG_DOMAIN @@ -59,7 +61,7 @@ mpc_seek_cb(cb_first_arg, mpc_int32_t offset) { struct mpc_decoder_data *data = (struct mpc_decoder_data *) cb_data; - return input_stream_seek(data->is, offset, SEEK_SET); + return input_stream_seek(data->is, offset, SEEK_SET, NULL); } static mpc_int32_t @@ -141,6 +143,7 @@ mpcdec_decode(struct decoder *mpd_decoder, struct input_stream *is) #endif mpc_reader reader; mpc_streaminfo info; + GError *error = NULL; struct audio_format audio_format; struct mpc_decoder_data data; @@ -150,11 +153,8 @@ mpcdec_decode(struct decoder *mpd_decoder, struct input_stream *is) mpc_uint32_t ret; int32_t chunk[G_N_ELEMENTS(sample_buffer)]; long bit_rate = 0; - unsigned long sample_pos = 0; mpc_uint32_t vbr_update_acc; mpc_uint32_t vbr_update_bits; - float total_time; - struct replay_gain_info *replay_gain_info = NULL; enum decoder_command cmd = DECODE_COMMAND_NONE; data.is = is; @@ -194,50 +194,47 @@ mpcdec_decode(struct decoder *mpd_decoder, struct input_stream *is) mpc_demux_get_info(demux, &info); #endif - audio_format.bits = 24; - audio_format.channels = info.channels; - audio_format.sample_rate = info.sample_freq; - - if (!audio_format_valid(&audio_format)) { + if (!audio_format_init_checked(&audio_format, info.sample_freq, + SAMPLE_FORMAT_S24_P32, + info.channels, &error)) { + g_warning("%s", error->message); + g_error_free(error); #ifndef MPC_IS_OLD_API mpc_demux_exit(demux); #endif - g_warning("Invalid audio format: %u:%u:%u\n", - audio_format.sample_rate, - audio_format.bits, - audio_format.channels); return; } - replay_gain_info = replay_gain_info_new(); + struct replay_gain_info replay_gain_info; + replay_gain_info_init(&replay_gain_info); #ifdef MPC_IS_OLD_API - replay_gain_info->tuples[REPLAY_GAIN_ALBUM].gain = info.gain_album * 0.01; - replay_gain_info->tuples[REPLAY_GAIN_ALBUM].peak = info.peak_album / 32767.0; - replay_gain_info->tuples[REPLAY_GAIN_TRACK].gain = info.gain_title * 0.01; - replay_gain_info->tuples[REPLAY_GAIN_TRACK].peak = info.peak_title / 32767.0; + replay_gain_info.tuples[REPLAY_GAIN_ALBUM].gain = info.gain_album * 0.01; + replay_gain_info.tuples[REPLAY_GAIN_ALBUM].peak = info.peak_album / 32767.0; + replay_gain_info.tuples[REPLAY_GAIN_TRACK].gain = info.gain_title * 0.01; + replay_gain_info.tuples[REPLAY_GAIN_TRACK].peak = info.peak_title / 32767.0; #else - replay_gain_info->tuples[REPLAY_GAIN_ALBUM].gain = MPC_OLD_GAIN_REF - (info.gain_album / 256.); - replay_gain_info->tuples[REPLAY_GAIN_ALBUM].peak = pow(10, info.peak_album / 256. / 20) / 32767; - replay_gain_info->tuples[REPLAY_GAIN_TRACK].gain = MPC_OLD_GAIN_REF - (info.gain_title / 256.); - replay_gain_info->tuples[REPLAY_GAIN_TRACK].peak = pow(10, info.peak_title / 256. / 20) / 32767; + replay_gain_info.tuples[REPLAY_GAIN_ALBUM].gain = MPC_OLD_GAIN_REF - (info.gain_album / 256.); + replay_gain_info.tuples[REPLAY_GAIN_ALBUM].peak = pow(10, info.peak_album / 256. / 20) / 32767; + replay_gain_info.tuples[REPLAY_GAIN_TRACK].gain = MPC_OLD_GAIN_REF - (info.gain_title / 256.); + replay_gain_info.tuples[REPLAY_GAIN_TRACK].peak = pow(10, info.peak_title / 256. / 20) / 32767; #endif + decoder_replay_gain(mpd_decoder, &replay_gain_info); + decoder_initialized(mpd_decoder, &audio_format, is->seekable, mpc_streaminfo_get_length(&info)); do { if (cmd == DECODE_COMMAND_SEEK) { - bool success; - - sample_pos = decoder_seek_where(mpd_decoder) * + mpc_int64_t where = decoder_seek_where(mpd_decoder) * audio_format.sample_rate; + bool success; #ifdef MPC_IS_OLD_API - success = mpc_decoder_seek_sample(&decoder, - sample_pos); + success = mpc_decoder_seek_sample(&decoder, where); #else - success = mpc_demux_seek_sample(demux, sample_pos) + success = mpc_demux_seek_sample(demux, where) == MPC_STATUS_OK; #endif if (success) @@ -268,33 +265,26 @@ mpcdec_decode(struct decoder *mpd_decoder, struct input_stream *is) ret = frame.samples; #endif - sample_pos += ret; - ret *= info.channels; mpc_to_mpd_buffer(chunk, sample_buffer, ret); - total_time = ((float)sample_pos) / audio_format.sample_rate; bit_rate = vbr_update_bits * audio_format.sample_rate / 1152 / 1000; cmd = decoder_data(mpd_decoder, is, chunk, ret * sizeof(chunk[0]), - total_time, - bit_rate, replay_gain_info); + bit_rate); } while (cmd != DECODE_COMMAND_STOP); - replay_gain_info_free(replay_gain_info); - #ifndef MPC_IS_OLD_API mpc_demux_exit(demux); #endif } static float -mpcdec_get_file_duration(const char *file) +mpcdec_get_file_duration(struct input_stream *is) { - struct input_stream is; float total_time = -1; mpc_reader reader; @@ -304,10 +294,7 @@ mpcdec_get_file_duration(const char *file) mpc_streaminfo info; struct mpc_decoder_data data; - if (!input_stream_open(&is, file)) - return -1; - - data.is = &is; + data.is = is; data.decoder = NULL; reader.read = mpc_read_cb; @@ -320,16 +307,12 @@ mpcdec_get_file_duration(const char *file) #ifdef MPC_IS_OLD_API mpc_streaminfo_init(&info); - if (mpc_streaminfo_read(&info, &reader) != ERROR_CODE_OK) { - input_stream_close(&is); + if (mpc_streaminfo_read(&info, &reader) != ERROR_CODE_OK) return -1; - } #else demux = mpc_demux_init(&reader); - if (demux == NULL) { - input_stream_close(&is); + if (demux == NULL) return -1; - } mpc_demux_get_info(demux, &info); mpc_demux_exit(demux); @@ -337,21 +320,17 @@ mpcdec_get_file_duration(const char *file) total_time = mpc_streaminfo_get_length(&info); - input_stream_close(&is); - return total_time; } static struct tag * -mpcdec_tag_dup(const char *file) +mpcdec_stream_tag(struct input_stream *is) { - float total_time = mpcdec_get_file_duration(file); + float total_time = mpcdec_get_file_duration(is); struct tag *tag; - if (total_time < 0) { - g_debug("Failed to get duration of file: %s", file); + if (total_time < 0) return NULL; - } tag = tag_new(); tag->time = total_time; @@ -363,6 +342,6 @@ static const char *const mpcdec_suffixes[] = { "mpc", NULL }; const struct decoder_plugin mpcdec_decoder_plugin = { .name = "mpcdec", .stream_decode = mpcdec_decode, - .tag_dup = mpcdec_tag_dup, + .stream_tag = mpcdec_stream_tag, .suffixes = mpcdec_suffixes, }; diff --git a/src/decoder/mpg123_decoder_plugin.c b/src/decoder/mpg123_decoder_plugin.c new file mode 100644 index 000000000..7b48ebfaf --- /dev/null +++ b/src/decoder/mpg123_decoder_plugin.c @@ -0,0 +1,209 @@ +/* + * Copyright (C) 2003-2010 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" /* must be first for large file support */ +#include "decoder_api.h" +#include "audio_check.h" + +#include <glib.h> + +#include <mpg123.h> + +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "mpg123" + +static bool +mpd_mpg123_init(G_GNUC_UNUSED const struct config_param *param) +{ + mpg123_init(); + + return true; +} + +static void +mpd_mpg123_finish(void) +{ + mpg123_exit(); +} + +/** + * Opens a file with an existing #mpg123_handle. + * + * @param handle a handle which was created before; on error, this + * function will not free it + * @param audio_format this parameter is filled after successful + * return + * @return true on success + */ +static bool +mpd_mpg123_open(mpg123_handle *handle, const char *path_fs, + struct audio_format *audio_format) +{ + GError *gerror = NULL; + char *path_dup; + int error; + int channels, encoding; + long rate; + + /* mpg123_open() wants a writable string :-( */ + path_dup = g_strdup(path_fs); + + error = mpg123_open(handle, path_dup); + g_free(path_dup); + if (error != MPG123_OK) { + g_warning("libmpg123 failed to open %s: %s", + path_fs, mpg123_plain_strerror(error)); + return false; + } + + /* obtain the audio format */ + + error = mpg123_getformat(handle, &rate, &channels, &encoding); + if (error != MPG123_OK) { + g_warning("mpg123_getformat() failed: %s", + mpg123_plain_strerror(error)); + return false; + } + + if (encoding != MPG123_ENC_SIGNED_16) { + /* other formats not yet implemented */ + g_warning("expected MPG123_ENC_SIGNED_16, got %d", encoding); + return false; + } + + if (!audio_format_init_checked(audio_format, rate, SAMPLE_FORMAT_S16, + channels, &gerror)) { + g_warning("%s", gerror->message); + g_error_free(gerror); + return false; + } + + return true; +} + +static void +mpd_mpg123_file_decode(struct decoder *decoder, const char *path_fs) +{ + struct audio_format audio_format; + mpg123_handle *handle; + int error; + off_t num_samples; + enum decoder_command cmd; + + /* open the file */ + + handle = mpg123_new(NULL, &error); + if (handle == NULL) { + g_warning("mpg123_new() failed: %s", + mpg123_plain_strerror(error)); + return; + } + + if (!mpd_mpg123_open(handle, path_fs, &audio_format)) { + mpg123_delete(handle); + return; + } + + num_samples = mpg123_length(handle); + + /* tell MPD core we're ready */ + + decoder_initialized(decoder, &audio_format, false, + (float)num_samples / + (float)audio_format.sample_rate); + + /* the decoder main loop */ + + do { + unsigned char buffer[8192]; + size_t nbytes; + + /* decode */ + + error = mpg123_read(handle, buffer, sizeof(buffer), &nbytes); + if (error != MPG123_OK) { + if (error != MPG123_DONE) + g_warning("mpg123_read() failed: %s", + mpg123_plain_strerror(error)); + break; + } + + /* send to MPD */ + + cmd = decoder_data(decoder, NULL, buffer, nbytes, 0); + + /* seeking not yet implemented */ + } while (cmd == DECODE_COMMAND_NONE); + + /* cleanup */ + + mpg123_delete(handle); +} + +static struct tag * +mpd_mpg123_tag_dup(const char *path_fs) +{ + struct audio_format audio_format; + mpg123_handle *handle; + int error; + off_t num_samples; + struct tag *tag; + + handle = mpg123_new(NULL, &error); + if (handle == NULL) { + g_warning("mpg123_new() failed: %s", + mpg123_plain_strerror(error)); + return NULL; + } + + if (!mpd_mpg123_open(handle, path_fs, &audio_format)) { + mpg123_delete(handle); + return NULL; + } + + num_samples = mpg123_length(handle); + if (num_samples <= 0) { + mpg123_delete(handle); + return NULL; + } + + tag = tag_new(); + + tag->time = num_samples / audio_format.sample_rate; + + /* ID3 tag support not yet implemented */ + + mpg123_delete(handle); + return tag; +} + +static const char *const mpg123_suffixes[] = { + "mp3", + NULL +}; + +const struct decoder_plugin mpg123_decoder_plugin = { + .name = "mpg123", + .init = mpd_mpg123_init, + .finish = mpd_mpg123_finish, + .file_decode = mpd_mpg123_file_decode, + /* streaming not yet implemented */ + .tag_dup = mpd_mpg123_tag_dup, + .suffixes = mpg123_suffixes, +}; diff --git a/src/decoder/oggflac_plugin.c b/src/decoder/oggflac_decoder_plugin.c index bdd589ccb..7e5f48318 100644 --- a/src/decoder/oggflac_plugin.c +++ b/src/decoder/oggflac_decoder_plugin.c @@ -1,5 +1,5 @@ /* - * Copyright (C) 2003-2009 The Music Player Daemon Project + * Copyright (C) 2003-2010 The Music Player Daemon Project * http://www.musicpd.org * * This program is free software; you can redistribute it and/or modify @@ -21,19 +21,18 @@ * OggFLAC support (half-stolen from flac_plugin.c :)) */ +#include "config.h" /* must be first for large file support */ #include "_flac_common.h" #include "_ogg_common.h" +#include "flac_metadata.h" #include <glib.h> #include <OggFLAC/seekable_stream_decoder.h> #include <assert.h> #include <unistd.h> -static void oggflac_cleanup(struct flac_data *data, - OggFLAC__SeekableStreamDecoder * decoder) +static void oggflac_cleanup(OggFLAC__SeekableStreamDecoder * decoder) { - if (data->replay_gain_info) - replay_gain_info_free(data->replay_gain_info); if (decoder) OggFLAC__seekable_stream_decoder_delete(decoder); } @@ -67,7 +66,7 @@ static OggFLAC__SeekableStreamDecoderSeekStatus of_seek_cb(G_GNUC_UNUSED const { struct flac_data *data = (struct flac_data *) fdata; - if (!input_stream_seek(data->input_stream, offset, SEEK_SET)) + if (!input_stream_seek(data->input_stream, offset, SEEK_SET, NULL)) return OggFLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_ERROR; return OggFLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_OK; @@ -156,13 +155,8 @@ oggflac_write_cb(G_GNUC_UNUSED const OggFLAC__SeekableStreamDecoder *decoder, void *vdata) { struct flac_data *data = (struct flac_data *) vdata; - FLAC__uint32 samples = frame->header.blocksize; - float time_change; - time_change = ((float)samples) / frame->header.sample_rate; - data->time += time_change; - - return flac_common_write(data, frame, buf); + return flac_common_write(data, frame, buf, 0); } /* used by TagDup */ @@ -173,17 +167,7 @@ static void of_metadata_dup_cb(G_GNUC_UNUSED const OggFLAC__SeekableStreamDecode assert(data->tag != NULL); - switch (block->type) { - case FLAC__METADATA_TYPE_STREAMINFO: - data->tag->time = ((float)block->data.stream_info. - total_samples) / - block->data.stream_info.sample_rate + 0.5; - return; - case FLAC__METADATA_TYPE_VORBIS_COMMENT: - flac_vorbis_comments_to_tag(data->tag, NULL, block); - default: - break; - } + flac_tag_apply_metadata(data->tag, NULL, block); } /* used by decode */ @@ -259,24 +243,20 @@ fail: /* public functions: */ static struct tag * -oggflac_tag_dup(const char *file) +oggflac_stream_tag(struct input_stream *is) { - struct input_stream input_stream; OggFLAC__SeekableStreamDecoder *decoder; struct flac_data data; + struct tag *tag; - if (!input_stream_open(&input_stream, file)) - return NULL; - if (ogg_stream_type_detect(&input_stream) != FLAC) { - input_stream_close(&input_stream); + if (ogg_stream_type_detect(is) != FLAC) return NULL; - } /* rewind the stream, because ogg_stream_type_detect() has moved it */ - input_stream_seek(&input_stream, 0, SEEK_SET); + input_stream_seek(is, 0, SEEK_SET, NULL); - flac_data_init(&data, NULL, &input_stream); + flac_data_init(&data, NULL, is); data.tag = tag_new(); @@ -284,15 +264,17 @@ oggflac_tag_dup(const char *file) * data.tag will be set or unset, that's all we care about */ decoder = full_decoder_init_and_read_metadata(&data, 1); - oggflac_cleanup(&data, decoder); - input_stream_close(&input_stream); + oggflac_cleanup(decoder); - if (!tag_is_defined(data.tag)) { - tag_free(data.tag); + if (tag_is_defined(data.tag)) { + tag = data.tag; data.tag = NULL; - } + } else + tag = NULL; - return data.tag; + flac_data_deinit(&data); + + return tag; } static void @@ -300,13 +282,14 @@ oggflac_decode(struct decoder * mpd_decoder, struct input_stream *input_stream) { OggFLAC__SeekableStreamDecoder *decoder = NULL; struct flac_data data; + struct audio_format audio_format; if (ogg_stream_type_detect(input_stream) != FLAC) return; /* rewind the stream, because ogg_stream_type_detect() has moved it */ - input_stream_seek(input_stream, 0, SEEK_SET); + input_stream_seek(input_stream, 0, SEEK_SET, NULL); flac_data_init(&data, mpd_decoder, input_stream); @@ -314,16 +297,13 @@ oggflac_decode(struct decoder * mpd_decoder, struct input_stream *input_stream) goto fail; } - if (!audio_format_valid(&data.audio_format)) { - g_warning("Invalid audio format: %u:%u:%u\n", - data.audio_format.sample_rate, - data.audio_format.bits, - data.audio_format.channels); + if (!data.initialized) goto fail; - } - decoder_initialized(mpd_decoder, &data.audio_format, - input_stream->seekable, data.total_time); + decoder_initialized(mpd_decoder, &audio_format, + input_stream->seekable, + (float)data.total_frames / + (float)data.audio_format.sample_rate); while (true) { OggFLAC__seekable_stream_decoder_process_single(decoder); @@ -333,11 +313,10 @@ oggflac_decode(struct decoder * mpd_decoder, struct input_stream *input_stream) } if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK) { FLAC__uint64 seek_sample = decoder_seek_where(mpd_decoder) * - data.audio_format.sample_rate + 0.5; + data.audio_format.sample_rate; if (OggFLAC__seekable_stream_decoder_seek_absolute (decoder, seek_sample)) { - data.time = ((float)seek_sample) / - data.audio_format.sample_rate; + data.next_frame = seek_sample; data.position = 0; decoder_command_finished(mpd_decoder); } else @@ -352,7 +331,8 @@ oggflac_decode(struct decoder * mpd_decoder, struct input_stream *input_stream) } fail: - oggflac_cleanup(&data, decoder); + oggflac_cleanup(decoder); + flac_data_deinit(&data); } static const char *const oggflac_suffixes[] = { "ogg", "oga", NULL }; @@ -368,7 +348,7 @@ static const char *const oggflac_mime_types[] = { const struct decoder_plugin oggflac_decoder_plugin = { .name = "oggflac", .stream_decode = oggflac_decode, - .tag_dup = oggflac_tag_dup, + .stream_tag = oggflac_stream_tag, .suffixes = oggflac_suffixes, .mime_types = oggflac_mime_types }; diff --git a/src/decoder/sidplay_decoder_plugin.cxx b/src/decoder/sidplay_decoder_plugin.cxx new file mode 100644 index 000000000..6fceeb30f --- /dev/null +++ b/src/decoder/sidplay_decoder_plugin.cxx @@ -0,0 +1,429 @@ +/* + * Copyright (C) 2003-2010 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" + +extern "C" { +#include "../decoder_api.h" +} + +#include <errno.h> +#include <stdlib.h> +#include <glib.h> + +#include <sidplay/sidplay2.h> +#include <sidplay/builders/resid.h> +#include <sidplay/utils/SidTuneMod.h> + +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "sidplay" + +#define SUBTUNE_PREFIX "tune_" + +static GPatternSpec *path_with_subtune; +static const char *songlength_file; +static GKeyFile *songlength_database; + +static bool all_files_are_containers; +static unsigned default_songlength; + +static bool filter_setting; + +static GKeyFile * +sidplay_load_songlength_db(const char *path) +{ + GError *error = NULL; + gchar *data; + gsize size; + + if (!g_file_get_contents(path, &data, &size, &error)) { + g_warning("unable to read songlengths file %s: %s", + path, error->message); + g_error_free(error); + return NULL; + } + + /* replace any ; comment characters with # */ + for (gsize i = 0; i < size; i++) + if (data[i] == ';') + data[i] = '#'; + + GKeyFile *db = g_key_file_new(); + bool success = g_key_file_load_from_data(db, data, size, + G_KEY_FILE_NONE, &error); + g_free(data); + if (!success) { + g_warning("unable to parse songlengths file %s: %s", + path, error->message); + g_error_free(error); + g_key_file_free(db); + return NULL; + } + + g_key_file_set_list_separator(db, ' '); + return db; +} + +static bool +sidplay_init(const struct config_param *param) +{ + /* read the songlengths database file */ + songlength_file=config_get_block_string(param, + "songlength_database", NULL); + if (songlength_file != NULL) + songlength_database = sidplay_load_songlength_db(songlength_file); + + default_songlength=config_get_block_unsigned(param, + "default_songlength", 0); + + all_files_are_containers=config_get_block_bool(param, + "all_files_are_containers", true); + + path_with_subtune=g_pattern_spec_new( + "*/" SUBTUNE_PREFIX "???.sid"); + + filter_setting=config_get_block_bool(param, "filter", true); + + return true; +} + +void +sidplay_finish() +{ + g_pattern_spec_free(path_with_subtune); + + if(songlength_database) + g_key_file_free(songlength_database); +} + +/** + * returns the file path stripped of any /tune_xxx.sid subtune + * suffix + */ +static char * +get_container_name(const char *path_fs) +{ + char *path_container=g_strdup(path_fs); + + if(!g_pattern_match(path_with_subtune, + strlen(path_container), path_container, NULL)) + return path_container; + + char *ptr=g_strrstr(path_container, "/" SUBTUNE_PREFIX); + if(ptr) *ptr='\0'; + + return path_container; +} + +/** + * returns tune number from file.sid/tune_xxx.sid style path or 1 if + * no subtune is appended + */ +static int +get_song_num(const char *path_fs) +{ + if(g_pattern_match(path_with_subtune, + strlen(path_fs), path_fs, NULL)) { + char *sub=g_strrstr(path_fs, "/" SUBTUNE_PREFIX); + if(!sub) return 1; + + sub+=strlen("/" SUBTUNE_PREFIX); + int song_num=strtol(sub, NULL, 10); + + if (errno == EINVAL) + return 1; + else + return song_num; + } else + return 1; +} + +/* get the song length in seconds */ +static int +get_song_length(const char *path_fs) +{ + if (songlength_database == NULL) + return -1; + + gchar *sid_file=get_container_name(path_fs); + SidTuneMod tune(sid_file); + g_free(sid_file); + if(!tune) { + g_warning("failed to load file for calculating md5 sum"); + return -1; + } + char md5sum[SIDTUNE_MD5_LENGTH+1]; + tune.createMD5(md5sum); + + int song_num=get_song_num(path_fs); + + gsize num_items; + gchar **values=g_key_file_get_string_list(songlength_database, + "Database", md5sum, &num_items, NULL); + if(!values || song_num>num_items) { + g_strfreev(values); + return -1; + } + + int minutes=strtol(values[song_num-1], NULL, 10); + if(errno==EINVAL) minutes=0; + + int seconds; + char *ptr=strchr(values[song_num-1], ':'); + if(ptr) { + seconds=strtol(ptr+1, NULL, 10); + if(errno==EINVAL) seconds=0; + } else + seconds=0; + + g_strfreev(values); + + return (minutes*60)+seconds; +} + +static void +sidplay_file_decode(struct decoder *decoder, const char *path_fs) +{ + int ret; + int channels; + + /* load the tune */ + + char *path_container=get_container_name(path_fs); + SidTune tune(path_container, NULL, true); + g_free(path_container); + if (!tune) { + g_warning("failed to load file"); + return; + } + + int song_num=get_song_num(path_fs); + tune.selectSong(song_num); + + int song_len=get_song_length(path_fs); + if(song_len==-1) song_len=default_songlength; + + /* initialize the player */ + + sidplay2 player; + int iret = player.load(&tune); + if (iret != 0) { + g_warning("sidplay2.load() failed: %s", player.error()); + return; + } + + /* initialize the builder */ + + ReSIDBuilder builder("ReSID"); + if (!builder) { + g_warning("failed to initialize ReSIDBuilder"); + return; + } + + builder.create(player.info().maxsids); + if (!builder) { + g_warning("ReSIDBuilder.create() failed"); + return; + } + + builder.filter(filter_setting); + if (!builder) { + g_warning("ReSIDBuilder.filter() failed"); + return; + } + + /* configure the player */ + + sid2_config_t config = player.config(); + + config.clockDefault = SID2_CLOCK_PAL; + config.clockForced = true; + config.clockSpeed = SID2_CLOCK_CORRECT; + config.frequency = 48000; + config.optimisation = SID2_DEFAULT_OPTIMISATION; + + config.precision = 16; + config.sidDefault = SID2_MOS6581; + config.sidEmulation = &builder; + config.sidModel = SID2_MODEL_CORRECT; + config.sidSamples = true; +#if G_BYTE_ORDER == G_LITTLE_ENDIAN + config.sampleFormat = SID2_LITTLE_SIGNED; +#else + config.sampleFormat = SID2_BIG_SIGNED; +#endif + if (tune.isStereo()) { + config.playback = sid2_stereo; + channels = 2; + } else { + config.playback = sid2_mono; + channels = 1; + } + + iret = player.config(config); + if (iret != 0) { + g_warning("sidplay2.config() failed: %s", player.error()); + return; + } + + /* initialize the MPD decoder */ + + struct audio_format audio_format; + audio_format_init(&audio_format, 48000, SAMPLE_FORMAT_S16, channels); + assert(audio_format_valid(&audio_format)); + + decoder_initialized(decoder, &audio_format, true, (float)song_len); + + /* .. and play */ + + const unsigned timebase = player.timebase(); + song_len *= timebase; + + enum decoder_command cmd; + do { + char buffer[4096]; + size_t nbytes; + + nbytes = player.play(buffer, sizeof(buffer)); + if (nbytes == 0) + break; + + decoder_timestamp(decoder, (double)player.time() / timebase); + + cmd = decoder_data(decoder, NULL, buffer, nbytes, 0); + + if(cmd==DECODE_COMMAND_SEEK) { + unsigned data_time = player.time(); + unsigned target_time = (unsigned) + (decoder_seek_where(decoder) * timebase); + + /* can't rewind so return to zero and seek forward */ + if(target_time<data_time) { + player.stop(); + data_time=0; + } + + /* ignore data until target time is reached */ + while(data_time<target_time) { + nbytes=player.play(buffer, sizeof(buffer)); + if(nbytes==0) + break; + data_time = player.time(); + } + + decoder_command_finished(decoder); + } + + if (song_len > 0 && player.time() >= song_len) + break; + + } while (cmd != DECODE_COMMAND_STOP); +} + +static struct tag * +sidplay_tag_dup(const char *path_fs) +{ + int song_num=get_song_num(path_fs); + char *path_container=get_container_name(path_fs); + + SidTune tune(path_container, NULL, true); + g_free(path_container); + if (!tune) + return NULL; + + const SidTuneInfo &info = tune.getInfo(); + struct tag *tag = tag_new(); + + /* title */ + const char *title; + if (info.numberOfInfoStrings > 0 && info.infoString[0] != NULL) + title=info.infoString[0]; + else + title=""; + + if(info.songs>1) { + char *tag_title=g_strdup_printf("%s (%d/%d)", + title, song_num, info.songs); + tag_add_item(tag, TAG_TITLE, tag_title); + g_free(tag_title); + } else + tag_add_item(tag, TAG_TITLE, title); + + /* artist */ + if (info.numberOfInfoStrings > 1 && info.infoString[1] != NULL) + tag_add_item(tag, TAG_ARTIST, info.infoString[1]); + + /* track */ + char *track=g_strdup_printf("%d", song_num); + tag_add_item(tag, TAG_TRACK, track); + g_free(track); + + /* time */ + int song_len=get_song_length(path_fs); + if(song_len!=-1) tag->time=song_len; + + return tag; +} + +static char * +sidplay_container_scan(const char *path_fs, const unsigned int tnum) +{ + SidTune tune(path_fs, NULL, true); + if (!tune) + return NULL; + + const SidTuneInfo &info=tune.getInfo(); + + /* Don't treat sids containing a single tune + as containers */ + if(!all_files_are_containers && info.songs<2) + return NULL; + + /* Construct container/tune path names, eg. + Delta.sid/tune_001.sid */ + if(tnum<=info.songs) { + char *subtune= g_strdup_printf( + SUBTUNE_PREFIX "%03u.sid", tnum); + return subtune; + } else + return NULL; +} + +static const char *const sidplay_suffixes[] = { + "sid", + "mus", + "str", + "prg", + "P00", + NULL +}; + +extern const struct decoder_plugin sidplay_decoder_plugin; +const struct decoder_plugin sidplay_decoder_plugin = { + "sidplay", + sidplay_init, + sidplay_finish, + NULL, /* stream_decode() */ + sidplay_file_decode, + sidplay_tag_dup, + NULL, /* stream_tag() */ + sidplay_container_scan, + sidplay_suffixes, + NULL, /* mime_types */ +}; diff --git a/src/decoder/sidplay_plugin.cxx b/src/decoder/sidplay_plugin.cxx deleted file mode 100644 index c62e6b4b6..000000000 --- a/src/decoder/sidplay_plugin.cxx +++ /dev/null @@ -1,163 +0,0 @@ -/* - * 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. - */ - -extern "C" { -#include "../decoder_api.h" -} - -#include <glib.h> - -#include <sidplay/sidplay2.h> -#include <sidplay/builders/resid.h> - -#undef G_LOG_DOMAIN -#define G_LOG_DOMAIN "sidplay" - -static void -sidplay_file_decode(struct decoder *decoder, const char *path_fs) -{ - int ret; - - /* load the tune */ - - SidTune tune(path_fs, NULL, true); - if (!tune) { - g_warning("failed to load file"); - return; - } - - tune.selectSong(1); - - /* initialize the player */ - - sidplay2 player; - int iret = player.load(&tune); - if (iret != 0) { - g_warning("sidplay2.load() failed: %s", player.error()); - return; - } - - /* initialize the builder */ - - ReSIDBuilder builder("ReSID"); - if (!builder) { - g_warning("failed to initialize ReSIDBuilder"); - return; - } - - builder.create(player.info().maxsids); - if (!builder) { - g_warning("ReSIDBuilder.create() failed"); - return; - } - - builder.filter(false); - if (!builder) { - g_warning("ReSIDBuilder.filter() failed"); - return; - } - - /* configure the player */ - - sid2_config_t config = player.config(); - - config.clockDefault = SID2_CLOCK_PAL; - config.clockForced = true; - config.clockSpeed = SID2_CLOCK_CORRECT; - config.frequency = 48000; - config.optimisation = SID2_DEFAULT_OPTIMISATION; - config.playback = sid2_stereo; - config.precision = 16; - config.sidDefault = SID2_MOS6581; - config.sidEmulation = &builder; - config.sidModel = SID2_MODEL_CORRECT; - config.sidSamples = true; -#if G_BYTE_ORDER == G_LITTLE_ENDIAN - config.sampleFormat = SID2_LITTLE_SIGNED; -#else - config.sampleFormat = SID2_BIG_SIGNED; -#endif - - iret = player.config(config); - if (iret != 0) { - g_warning("sidplay2.config() failed: %s", player.error()); - return; - } - - /* initialize the MPD decoder */ - - struct audio_format audio_format; - audio_format.sample_rate = 48000; - audio_format.bits = 16; - audio_format.channels = 2; - - decoder_initialized(decoder, &audio_format, false, -1); - - /* .. and play */ - - enum decoder_command cmd; - do { - char buffer[4096]; - size_t nbytes; - - nbytes = player.play(buffer, sizeof(buffer)); - if (nbytes == 0) - break; - - cmd = decoder_data(decoder, NULL, buffer, nbytes, - 0, 0, NULL); - } while (cmd == DECODE_COMMAND_NONE); -} - -static struct tag * -sidplay_tag_dup(const char *path_fs) -{ - SidTune tune(path_fs, NULL, true); - if (!tune) - return NULL; - - const SidTuneInfo &info = tune.getInfo(); - struct tag *tag = tag_new(); - - if (info.numberOfInfoStrings > 0 && info.infoString[0] != NULL) - tag_add_item(tag, TAG_ITEM_TITLE, info.infoString[0]); - - if (info.numberOfInfoStrings > 1 && info.infoString[1] != NULL) - tag_add_item(tag, TAG_ITEM_ARTIST, info.infoString[1]); - - return tag; -} - -static const char *const sidplay_suffixes[] = { - "sid", - NULL -}; - -extern const struct decoder_plugin sidplay_decoder_plugin; -const struct decoder_plugin sidplay_decoder_plugin = { - "sidplay", - NULL, /* init() */ - NULL, /* finish() */ - NULL, /* stream_decode() */ - sidplay_file_decode, - sidplay_tag_dup, - NULL, /* container_scan */ - sidplay_suffixes, - NULL, /* mime_types */ -}; diff --git a/src/decoder/sndfile_decoder_plugin.c b/src/decoder/sndfile_decoder_plugin.c new file mode 100644 index 000000000..af68f117d --- /dev/null +++ b/src/decoder/sndfile_decoder_plugin.c @@ -0,0 +1,251 @@ +/* + * Copyright (C) 2003-2010 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 <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, NULL); + 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; + GError *error = NULL; + size_t nbytes; + + nbytes = input_stream_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 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) +{ + 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 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_TITLE, p); + + p = sf_get_string(sf, SF_STR_ARTIST); + if (p != NULL) + tag_add_item(tag, TAG_ARTIST, p); + + p = sf_get_string(sf, SF_STR_DATE); + if (p != NULL) + tag_add_item(tag, TAG_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, +}; diff --git a/src/decoder/vorbis_plugin.c b/src/decoder/vorbis_decoder_plugin.c index 7c782a779..0a3944ad6 100644 --- a/src/decoder/vorbis_plugin.c +++ b/src/decoder/vorbis_decoder_plugin.c @@ -1,5 +1,5 @@ /* - * Copyright (C) 2003-2009 The Music Player Daemon Project + * Copyright (C) 2003-2010 The Music Player Daemon Project * http://www.musicpd.org * * This program is free software; you can redistribute it and/or modify @@ -17,19 +17,19 @@ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ -/* TODO 'ogg' should probably be replaced with 'oggvorbis' in all instances */ - -#include "_ogg_common.h" #include "config.h" +#include "_ogg_common.h" +#include "audio_check.h" #include "uri.h" #ifndef HAVE_TREMOR +#define OV_EXCLUDE_STATIC_CALLBACKS #include <vorbis/vorbisfile.h> #else #include <tremor/ivorbisfile.h> /* Macros to make Tremor's API look like libogg. Tremor always returns host-byte-order 16-bit signed data, and uses integer - milliseconds where libogg uses double seconds. + milliseconds where libogg uses double seconds. */ #define ov_read(VF, BUFFER, LENGTH, BIGENDIANP, WORD, SGNED, BITSTREAM) \ ov_read(VF, BUFFER, LENGTH, BITSTREAM) @@ -55,46 +55,46 @@ #define OGG_DECODE_USE_BIGENDIAN 0 #endif -typedef struct _OggCallbackData { +struct vorbis_input_stream { struct decoder *decoder; struct input_stream *input_stream; bool seekable; -} OggCallbackData; +}; -static size_t ogg_read_cb(void *ptr, size_t size, size_t nmemb, void *vdata) +static size_t ogg_read_cb(void *ptr, size_t size, size_t nmemb, void *data) { + struct vorbis_input_stream *vis = data; size_t ret; - OggCallbackData *data = (OggCallbackData *) vdata; - ret = decoder_read(data->decoder, data->input_stream, ptr, size * nmemb); + ret = decoder_read(vis->decoder, vis->input_stream, ptr, size * nmemb); errno = 0; return ret / size; } -static int ogg_seek_cb(void *vdata, ogg_int64_t offset, int whence) +static int ogg_seek_cb(void *data, ogg_int64_t offset, int whence) { - const OggCallbackData *data = (const OggCallbackData *) vdata; + struct vorbis_input_stream *vis = data; - return data->seekable && - decoder_get_command(data->decoder) != DECODE_COMMAND_STOP && - input_stream_seek(data->input_stream, offset, whence) + return vis->seekable && + (!vis->decoder || decoder_get_command(vis->decoder) != DECODE_COMMAND_STOP) && + input_stream_seek(vis->input_stream, offset, whence, NULL) ? 0 : -1; } /* TODO: check Ogg libraries API and see if we can just not have this func */ -static int ogg_close_cb(G_GNUC_UNUSED void *vdata) +static int ogg_close_cb(G_GNUC_UNUSED void *data) { return 0; } -static long ogg_tell_cb(void *vdata) +static long ogg_tell_cb(void *data) { - const OggCallbackData *data = (const OggCallbackData *) vdata; + const struct vorbis_input_stream *vis = data; - return (long)data->input_stream->offset; + return (long)vis->input_stream->offset; } static const ov_callbacks vorbis_is_callbacks = { @@ -105,6 +105,52 @@ static const ov_callbacks vorbis_is_callbacks = { }; static const char * +vorbis_strerror(int code) +{ + switch (code) { + case OV_EREAD: + return "read error"; + + case OV_ENOTVORBIS: + return "not vorbis stream"; + + case OV_EVERSION: + return "vorbis version mismatch"; + + case OV_EBADHEADER: + return "invalid vorbis header"; + + case OV_EFAULT: + return "internal logic error"; + + default: + return "unknown error"; + } +} + +static bool +vorbis_is_open(struct vorbis_input_stream *vis, OggVorbis_File *vf, + struct decoder *decoder, struct input_stream *input_stream) +{ + vis->decoder = decoder; + vis->input_stream = input_stream; + vis->seekable = input_stream->seekable && + (input_stream->uri == NULL || + !uri_has_scheme(input_stream->uri)); + + int ret = ov_open_callbacks(vis, vf, NULL, 0, vorbis_is_callbacks); + if (ret < 0) { + if (decoder == NULL || + decoder_get_command(decoder) == DECODE_COMMAND_NONE) + g_warning("Failed to open Ogg Vorbis stream: %s", + vorbis_strerror(ret)); + return false; + } + + return true; +} + +static const char * vorbis_comment_value(const char *comment, const char *needle) { size_t len = strlen(needle); @@ -116,14 +162,13 @@ vorbis_comment_value(const char *comment, const char *needle) return NULL; } -static struct replay_gain_info * -vorbis_comments_to_replay_gain(char **comments) +static bool +vorbis_comments_to_replay_gain(struct replay_gain_info *rgi, char **comments) { - struct replay_gain_info *rgi; const char *temp; bool found = false; - rgi = replay_gain_info_new(); + replay_gain_info_init(rgi); while (*comments) { if ((temp = @@ -147,12 +192,7 @@ vorbis_comments_to_replay_gain(char **comments) comments++; } - if (!found) { - replay_gain_info_free(rgi); - rgi = NULL; - } - - return rgi; + return found; } static const char *VORBIS_COMMENT_TRACK_KEY = "tracknumber"; @@ -183,11 +223,11 @@ vorbis_parse_comment(struct tag *tag, const char *comment) assert(tag != NULL); if (vorbis_copy_comment(tag, comment, VORBIS_COMMENT_TRACK_KEY, - TAG_ITEM_TRACK) || + TAG_TRACK) || vorbis_copy_comment(tag, comment, VORBIS_COMMENT_DISC_KEY, - TAG_ITEM_DISC) || + TAG_DISC) || vorbis_copy_comment(tag, comment, "album artist", - TAG_ITEM_ALBUM_ARTIST)) + TAG_ALBUM_ARTIST)) return; for (unsigned i = 0; i < TAG_NUM_OF_ITEM_TYPES; ++i) @@ -226,42 +266,23 @@ vorbis_send_comments(struct decoder *decoder, struct input_stream *is, tag_free(tag); } -static bool -oggvorbis_seekable(struct decoder *decoder) -{ - char *uri; - bool seekable; - - uri = decoder_get_uri(decoder); - if (uri == NULL) - return false; - - /* disable seeking on remote streams, because libvorbis seeks - around like crazy, and due to being very expensive, this - delays song playback my 10 or 20 seconds */ - seekable = !uri_has_scheme(uri); - g_free(uri); - - return seekable; -} - /* public */ static void vorbis_stream_decode(struct decoder *decoder, struct input_stream *input_stream) { + GError *error = NULL; OggVorbis_File vf; - OggCallbackData data; + struct vorbis_input_stream vis; struct audio_format audio_format; + float total_time; int current_section; int prev_section = -1; long ret; char chunk[OGG_CHUNK_SIZE]; long bitRate = 0; long test; - struct replay_gain_info *replay_gain_info = NULL; - char **comments; - bool initialized = false; + const vorbis_info *vi; enum decoder_command cmd = DECODE_COMMAND_NONE; if (ogg_stream_type_detect(input_stream) != VORBIS) @@ -269,43 +290,30 @@ vorbis_stream_decode(struct decoder *decoder, /* rewind the stream, because ogg_stream_type_detect() has moved it */ - input_stream_seek(input_stream, 0, SEEK_SET); - - data.decoder = decoder; - data.input_stream = input_stream; - data.seekable = input_stream->seekable && oggvorbis_seekable(decoder); + input_stream_seek(input_stream, 0, SEEK_SET, NULL); - if ((ret = ov_open_callbacks(&data, &vf, NULL, 0, - vorbis_is_callbacks)) < 0) { - const char *error; - if (decoder_get_command(decoder) != DECODE_COMMAND_NONE) - return; + if (!vorbis_is_open(&vis, &vf, decoder, input_stream)) + return; - switch (ret) { - case OV_EREAD: - error = "read error"; - break; - case OV_ENOTVORBIS: - error = "not vorbis stream"; - break; - case OV_EVERSION: - error = "vorbis version mismatch"; - break; - case OV_EBADHEADER: - error = "invalid vorbis header"; - break; - case OV_EFAULT: - error = "internal logic error"; - break; - default: - error = "unknown error"; - break; - } + vi = ov_info(&vf, -1); + if (vi == NULL) { + g_warning("ov_info() has failed"); + return; + } - g_warning("Error decoding Ogg Vorbis stream: %s", error); + if (!audio_format_init_checked(&audio_format, vi->rate, + SAMPLE_FORMAT_S16, + vi->channels, &error)) { + g_warning("%s", error->message); + g_error_free(error); return; } - audio_format.bits = 16; + + total_time = ov_time_total(&vf, -1); + if (total_time < 0) + total_time = 0; + + decoder_initialized(decoder, &audio_format, vis.seekable, total_time); do { if (cmd == DECODE_COMMAND_SEEK) { @@ -325,83 +333,61 @@ vorbis_stream_decode(struct decoder *decoder, break; if (current_section != prev_section) { - /*printf("new song!\n"); */ - vorbis_info *vi = ov_info(&vf, -1); - struct replay_gain_info *new_rgi; - - audio_format.channels = vi->channels; - audio_format.sample_rate = vi->rate; - - if (!audio_format_valid(&audio_format)) { - g_warning("Invalid audio format: %u:%u:%u\n", - audio_format.sample_rate, - audio_format.bits, - audio_format.channels); + char **comments; + + vi = ov_info(&vf, -1); + if (vi == NULL) { + g_warning("ov_info() has failed"); break; } - if (!initialized) { - float total_time = ov_time_total(&vf, -1); - if (total_time < 0) - total_time = 0; - decoder_initialized(decoder, &audio_format, - data.seekable, - total_time); - initialized = true; + if (vi->rate != (long)audio_format.sample_rate || + vi->channels != (int)audio_format.channels) { + /* we don't support audio format + change yet */ + g_warning("audio format change, stopping here"); + break; } + comments = ov_comment(&vf, -1)->user_comments; vorbis_send_comments(decoder, input_stream, comments); - new_rgi = vorbis_comments_to_replay_gain(comments); - if (new_rgi != NULL) { - if (replay_gain_info != NULL) - replay_gain_info_free(replay_gain_info); - replay_gain_info = new_rgi; - } - } - prev_section = current_section; + struct replay_gain_info rgi; + if (vorbis_comments_to_replay_gain(&rgi, comments)) + decoder_replay_gain(decoder, &rgi); + + prev_section = current_section; + } if ((test = ov_bitrate_instant(&vf)) > 0) bitRate = test / 1000; cmd = decoder_data(decoder, input_stream, chunk, ret, - ov_pcm_tell(&vf) / audio_format.sample_rate, - bitRate, replay_gain_info); + bitRate); } while (cmd != DECODE_COMMAND_STOP); - if (replay_gain_info) - replay_gain_info_free(replay_gain_info); - ov_clear(&vf); } static struct tag * -vorbis_tag_dup(const char *file) +vorbis_stream_tag(struct input_stream *is) { - struct tag *ret; - FILE *fp; + struct vorbis_input_stream vis; OggVorbis_File vf; - fp = fopen(file, "r"); - if (!fp) { + if (!vorbis_is_open(&vis, &vf, NULL, is)) return NULL; - } - if (ov_open(fp, &vf, NULL, 0) < 0) { - fclose(fp); - return NULL; - } - - ret = vorbis_comments_to_tag(ov_comment(&vf, -1)->user_comments); + struct tag *tag = vorbis_comments_to_tag(ov_comment(&vf, -1)->user_comments); - if (!ret) - ret = tag_new(); - ret->time = (int)(ov_time_total(&vf, -1) + 0.5); + if (tag == NULL) + tag = tag_new(); + tag->time = (int)(ov_time_total(&vf, -1) + 0.5); ov_clear(&vf); - return ret; + return tag; } static const char *const vorbis_suffixes[] = { @@ -423,7 +409,7 @@ static const char *const vorbis_mime_types[] = { const struct decoder_plugin vorbis_decoder_plugin = { .name = "vorbis", .stream_decode = vorbis_stream_decode, - .tag_dup = vorbis_tag_dup, + .stream_tag = vorbis_stream_tag, .suffixes = vorbis_suffixes, .mime_types = vorbis_mime_types }; diff --git a/src/decoder/wavpack_plugin.c b/src/decoder/wavpack_decoder_plugin.c index 7ad3a62b0..efed98851 100644 --- a/src/decoder/wavpack_plugin.c +++ b/src/decoder/wavpack_decoder_plugin.c @@ -1,5 +1,5 @@ /* - * Copyright (C) 2003-2009 The Music Player Daemon Project + * Copyright (C) 2003-2010 The Music Player Daemon Project * http://www.musicpd.org * * This program is free software; you can redistribute it and/or modify @@ -17,9 +17,11 @@ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ -#include "../decoder_api.h" -#include "../path.h" -#include "../utils.h" +#include "config.h" +#include "decoder_api.h" +#include "audio_check.h" +#include "path.h" +#include "utils.h" #include <wavpack/wavpack.h> #include <glib.h> @@ -41,17 +43,17 @@ static struct { const char *name; enum tag_type type; } tagtypes[] = { - { "artist", TAG_ITEM_ARTIST }, - { "album", TAG_ITEM_ALBUM }, - { "title", TAG_ITEM_TITLE }, - { "track", TAG_ITEM_TRACK }, - { "name", TAG_ITEM_NAME }, - { "genre", TAG_ITEM_GENRE }, - { "date", TAG_ITEM_DATE }, - { "composer", TAG_ITEM_COMPOSER }, - { "performer", TAG_ITEM_PERFORMER }, - { "comment", TAG_ITEM_COMMENT }, - { "disc", TAG_ITEM_DISC }, + { "artist", TAG_ARTIST }, + { "album", TAG_ALBUM }, + { "title", TAG_TITLE }, + { "track", TAG_TRACK }, + { "name", TAG_NAME }, + { "genre", TAG_GENRE }, + { "date", TAG_DATE }, + { "composer", TAG_COMPOSER }, + { "performer", TAG_PERFORMER }, + { "comment", TAG_COMMENT }, + { "disc", TAG_DISC }, }; /** A pointer type for format converter function. */ @@ -97,19 +99,11 @@ format_samples_int(int bytes_per_sample, void *buffer, uint32_t count) } break; } + case 3: + case 4: /* do nothing */ break; - case 4: { - uint32_t *dst = buffer; - assert_static(sizeof(*dst) <= sizeof(*src)); - - /* downsample to 24-bit */ - while (count--) { - *dst++ = *src++ >> 8; - } - break; - } } } @@ -129,38 +123,61 @@ format_samples_float(G_GNUC_UNUSED int bytes_per_sample, void *buffer, } } +/** + * Choose a MPD sample format from libwavpacks' number of bits. + */ +static enum sample_format +wavpack_bits_to_sample_format(bool is_float, int bytes_per_sample) +{ + if (is_float) + return SAMPLE_FORMAT_S24_P32; + + switch (bytes_per_sample) { + case 1: + return SAMPLE_FORMAT_S8; + + case 2: + return SAMPLE_FORMAT_S16; + + case 3: + return SAMPLE_FORMAT_S24_P32; + + case 4: + return SAMPLE_FORMAT_S32; + + default: + return SAMPLE_FORMAT_UNDEFINED; + } +} + /* * This does the main decoding thing. * Requires an already opened WavpackContext. */ static void -wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek, - struct replay_gain_info *replay_gain_info) +wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek) { + GError *error = NULL; + bool is_float; + enum sample_format sample_format; struct audio_format audio_format; format_samples_t format_samples; char chunk[CHUNK_SIZE]; int samples_requested, samples_got; - float total_time, current_time; + float total_time; int bytes_per_sample, output_sample_size; - int position; - audio_format.sample_rate = WavpackGetSampleRate(wpc); - audio_format.channels = WavpackGetReducedChannels(wpc); - audio_format.bits = WavpackGetBitsPerSample(wpc); - - /* round bitwidth to 8-bit units */ - audio_format.bits = (audio_format.bits + 7) & (~7); - /* mpd handles max 24-bit samples */ - if (audio_format.bits > 24) { - audio_format.bits = 24; - } - - if (!audio_format_valid(&audio_format)) { - g_warning("Invalid audio format: %u:%u:%u\n", - audio_format.sample_rate, - audio_format.bits, - audio_format.channels); + is_float = (WavpackGetMode(wpc) & MODE_FLOAT) != 0; + sample_format = + wavpack_bits_to_sample_format(is_float, + WavpackGetBytesPerSample(wpc)); + + if (!audio_format_init_checked(&audio_format, + WavpackGetSampleRate(wpc), + sample_format, + WavpackGetNumChannels(wpc), &error)) { + g_warning("%s", error->message); + g_error_free(error); return; } @@ -180,8 +197,6 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek, decoder_initialized(decoder, &audio_format, can_seek, total_time); - position = 0; - do { if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) { if (can_seek) { @@ -189,7 +204,6 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek, audio_format.sample_rate; if (WavpackSeekSample(wpc, where)) { - position = where; decoder_command_finished(decoder); } else { decoder_seek_error(decoder); @@ -209,9 +223,6 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek, if (samples_got > 0) { int bitrate = (int)(WavpackGetInstantBitrate(wpc) / 1000 + 0.5); - position += samples_got; - current_time = position; - current_time /= audio_format.sample_rate; format_samples( bytes_per_sample, chunk, @@ -221,8 +232,7 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek, decoder_data( decoder, NULL, chunk, samples_got * output_sample_size, - current_time, bitrate, - replay_gain_info + bitrate ); } } while (samples_got > 0); @@ -246,13 +256,13 @@ wavpack_tag_float(WavpackContext *wpc, const char *key, float *value_r) return true; } -static struct replay_gain_info * -wavpack_replaygain(WavpackContext *wpc) +static bool +wavpack_replaygain(struct replay_gain_info *replay_gain_info, + WavpackContext *wpc) { - struct replay_gain_info *replay_gain_info; bool found = false; - replay_gain_info = replay_gain_info_new(); + replay_gain_info_init(replay_gain_info); found |= wavpack_tag_float( wpc, "replaygain_track_gain", @@ -271,13 +281,7 @@ wavpack_replaygain(WavpackContext *wpc) &replay_gain_info->tuples[REPLAY_GAIN_ALBUM].peak ); - if (found) { - return replay_gain_info; - } - - replay_gain_info_free(replay_gain_info); - - return NULL; + return found; } /* @@ -397,13 +401,13 @@ wavpack_input_get_pos(void *id) static int wavpack_input_set_pos_abs(void *id, uint32_t pos) { - return input_stream_seek(wpin(id)->is, pos, SEEK_SET) ? 0 : -1; + return input_stream_seek(wpin(id)->is, pos, SEEK_SET, NULL) ? 0 : -1; } static int wavpack_input_set_pos_rel(void *id, int32_t delta, int mode) { - return input_stream_seek(wpin(id)->is, delta, mode) ? 0 : -1; + return input_stream_seek(wpin(id)->is, delta, mode, NULL) ? 0 : -1; } static int @@ -452,13 +456,12 @@ wavpack_input_init(struct wavpack_input *isp, struct decoder *decoder, isp->last_byte = EOF; } -static bool -wavpack_open_wvc(struct decoder *decoder, struct input_stream *is_wvc, +static struct input_stream * +wavpack_open_wvc(struct decoder *decoder, const char *uri, struct wavpack_input *wpi) { - char *utf8url; + struct input_stream *is_wvc; char *wvc_url = NULL; - bool ret; char first_byte; size_t nbytes; @@ -466,20 +469,15 @@ wavpack_open_wvc(struct decoder *decoder, struct input_stream *is_wvc, * As we use dc->utf8url, this function will be bad for * single files. utf8url is not absolute file path :/ */ - utf8url = decoder_get_uri(decoder); - if (utf8url == NULL) { + if (uri == NULL) return false; - } - wvc_url = g_strconcat(utf8url, "c", NULL); - g_free(utf8url); - - ret = input_stream_open(is_wvc, wvc_url); + wvc_url = g_strconcat(uri, "c", NULL); + is_wvc = input_stream_open(wvc_url, NULL); g_free(wvc_url); - if (!ret) { - return false; - } + if (is_wvc == NULL) + return NULL; /* * And we try to buffer in order to get know @@ -490,13 +488,13 @@ wavpack_open_wvc(struct decoder *decoder, struct input_stream *is_wvc, ); if (nbytes == 0) { input_stream_close(is_wvc); - return false; + return NULL; } /* push it back */ wavpack_input_init(wpi, decoder, is_wvc); wpi->last_byte = first_byte; - return true; + return is_wvc; } /* @@ -507,14 +505,15 @@ wavpack_streamdecode(struct decoder * decoder, struct input_stream *is) { char error[ERRORLEN]; WavpackContext *wpc; - struct input_stream is_wvc; - int open_flags = OPEN_2CH_MAX | OPEN_NORMALIZE; + struct input_stream *is_wvc; + int open_flags = OPEN_NORMALIZE; struct wavpack_input isp, isp_wvc; bool can_seek = is->seekable; - if (wavpack_open_wvc(decoder, &is_wvc, &isp_wvc)) { + is_wvc = wavpack_open_wvc(decoder, is->uri, &isp_wvc); + if (is_wvc != NULL) { open_flags |= OPEN_WVC; - can_seek &= is_wvc.seekable; + can_seek &= is_wvc->seekable; } if (!can_seek) { @@ -533,11 +532,11 @@ wavpack_streamdecode(struct decoder * decoder, struct input_stream *is) return; } - wavpack_decode(decoder, wpc, can_seek, NULL); + wavpack_decode(decoder, wpc, can_seek); WavpackCloseFile(wpc); if (open_flags & OPEN_WVC) { - input_stream_close(&is_wvc); + input_stream_close(is_wvc); } } @@ -549,11 +548,10 @@ wavpack_filedecode(struct decoder *decoder, const char *fname) { char error[ERRORLEN]; WavpackContext *wpc; - struct replay_gain_info *replay_gain_info; wpc = WavpackOpenFileInput( fname, error, - OPEN_TAGS | OPEN_WVC | OPEN_2CH_MAX | OPEN_NORMALIZE, 23 + OPEN_TAGS | OPEN_WVC | OPEN_NORMALIZE, 23 ); if (wpc == NULL) { g_warning( @@ -563,13 +561,11 @@ wavpack_filedecode(struct decoder *decoder, const char *fname) return; } - replay_gain_info = wavpack_replaygain(wpc); + struct replay_gain_info replay_gain_info; + if (wavpack_replaygain(&replay_gain_info, wpc)) + decoder_replay_gain(decoder, &replay_gain_info); - wavpack_decode(decoder, wpc, true, replay_gain_info); - - if (replay_gain_info) { - replay_gain_info_free(replay_gain_info); - } + wavpack_decode(decoder, wpc, true); WavpackCloseFile(wpc); } diff --git a/src/decoder/wildmidi_plugin.c b/src/decoder/wildmidi_decoder_plugin.c index b5e9810f9..66e6c61cf 100644 --- a/src/decoder/wildmidi_plugin.c +++ b/src/decoder/wildmidi_decoder_plugin.c @@ -1,5 +1,5 @@ /* - * Copyright (C) 2003-2009 The Music Player Daemon Project + * Copyright (C) 2003-2010 The Music Player Daemon Project * http://www.musicpd.org * * This program is free software; you can redistribute it and/or modify @@ -17,7 +17,8 @@ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ -#include "../decoder_api.h" +#include "config.h" +#include "decoder_api.h" #include <glib.h> @@ -58,7 +59,7 @@ wildmidi_file_decode(struct decoder *decoder, const char *path_fs) { static const struct audio_format audio_format = { .sample_rate = WILDMIDI_SAMPLE_RATE, - .bits = 16, + .format = SAMPLE_FORMAT_S16, .channels = 2, }; midi *wm; @@ -90,10 +91,7 @@ wildmidi_file_decode(struct decoder *decoder, const char *path_fs) if (len <= 0) break; - cmd = decoder_data(decoder, NULL, buffer, len, - (float)info->current_sample / - (float)WILDMIDI_SAMPLE_RATE, - 0, NULL); + cmd = decoder_data(decoder, NULL, buffer, len, 0); if (cmd == DECODE_COMMAND_SEEK) { unsigned long seek_where = WILDMIDI_SAMPLE_RATE * @@ -116,21 +114,17 @@ wildmidi_file_decode(struct decoder *decoder, const char *path_fs) static struct tag * wildmidi_tag_dup(const char *path_fs) { - midi *wm; - const struct _WM_Info *info; - struct tag *tag; - - wm = WildMidi_Open(path_fs); + midi *wm = WildMidi_Open(path_fs); if (wm == NULL) return NULL; - info = WildMidi_GetInfo(wm); + const struct _WM_Info *info = WildMidi_GetInfo(wm); if (info == NULL) { WildMidi_Close(wm); return NULL; } - tag = tag_new(); + struct tag *tag = tag_new(); tag->time = info->approx_total_samples / WILDMIDI_SAMPLE_RATE; WildMidi_Close(wm); |