diff options
Diffstat (limited to 'src/decoder')
26 files changed, 2086 insertions, 1166 deletions
diff --git a/src/decoder/_flac_common.c b/src/decoder/_flac_common.c index 7c8fe9875..4dd2b46ea 100644 --- a/src/decoder/_flac_common.c +++ b/src/decoder/_flac_common.c @@ -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 +35,96 @@ 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->have_stream_info = false; + 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) -{ - 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; -} - -static void -flac_parse_replay_gain(const FLAC__StreamMetadata *block, - struct flac_data *data) +void +flac_data_deinit(struct flac_data *data) { - bool found = false; + pcm_buffer_deinit(&data->buffer); - if (data->replay_gain_info) + if (data->replay_gain_info != NULL) 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(const FLAC__StreamMetadata_StreamInfo *si) { - size_t name_length = strlen(name); - size_t char_tnum_length = 0; - const char *comment = (const char*)entry->entry; + switch (si->bits_per_sample) { + case 8: + return SAMPLE_FORMAT_S8; - if (entry->length <= name_length || - g_ascii_strncasecmp(comment, name, name_length) != 0) - return NULL; + case 16: + return SAMPLE_FORMAT_S16; - 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; - } + case 24: + return SAMPLE_FORMAT_S24_P32; - if (comment[name_length] == '=') { - *length_r = entry->length - name_length - 1; - return comment + name_length + 1; - } + case 32: + return SAMPLE_FORMAT_S32; - return NULL; + default: + return SAMPLE_FORMAT_UNDEFINED; + } } -/** - * 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) +bool +flac_data_get_audio_format(struct flac_data *data, + struct audio_format *audio_format) { - const char *value; - size_t value_length; + GError *error = NULL; - 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; + if (!data->have_stream_info) { + g_warning("no STREAMINFO packet found"); + return false; } - 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"; + data->sample_format = flac_sample_format(&data->stream_info); -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_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)) - return; + if (!audio_format_init_checked(audio_format, + data->stream_info.sample_rate, + data->sample_format, + data->stream_info.channels, &error)) { + g_warning("%s", error->message); + g_error_free(error); + return false; + } - 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; -} + data->frame_size = audio_format_frame_size(audio_format); -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; - - for (unsigned i = block->data.vorbis_comment.num_comments; i > 0; --i) - flac_parse_comment(tag, char_tnum, comments++); + return true; } void flac_metadata_common_cb(const FLAC__StreamMetadata * block, struct flac_data *data) { - const FLAC__StreamMetadata_StreamInfo *si = &(block->data.stream_info); - 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); + data->stream_info = block->data.stream_info; + data->have_stream_info = true; break; + case FLAC__METADATA_TYPE_VORBIS_COMMENT: - flac_parse_replay_gain(block, data); + if (data->replay_gain_info) + replay_gain_info_free(data->replay_gain_info); + data->replay_gain_info = flac_parse_replay_gain(block); 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; @@ -239,133 +153,43 @@ 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! - */ -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) +FLAC__StreamDecoderWriteStatus +flac_common_write(struct flac_data *data, const FLAC__Frame * frame, + const FLAC__int32 *const buf[], + FLAC__uint64 nbytes) { - unsigned int c_chan; - - for (; position < end; ++position) - for (c_chan = 0; c_chan < num_channels; c_chan++) - *dest++ = buf[c_chan][position]; -} + enum decoder_command cmd; + size_t buffer_size = frame->header.blocksize * data->frame_size; + void *buffer; + unsigned bit_rate; -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; + buffer = pcm_buffer_get(&data->buffer, buffer_size); - case 4: - flac_convert_32((int32_t*)dest, num_channels, buf, - position, end); - break; + flac_convert(buffer, frame->header.channels, + data->sample_format, buf, + 0, frame->header.blocksize); - case 1: - flac_convert_8((int8_t*)dest, num_channels, buf, - position, end); + 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->replay_gain_info); + data->next_frame += frame->header.blocksize; + switch (cmd) { + case DECODE_COMMAND_NONE: + case DECODE_COMMAND_START: break; - } -} - -FLAC__StreamDecoderWriteStatus -flac_common_write(struct flac_data *data, const FLAC__Frame * frame, - const FLAC__int32 *const buf[]) -{ - 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; - if (bytes_per_sample != 1 && bytes_per_sample != 2 && - bytes_per_sample != 4) - /* exotic unsupported bit rate */ + case DECODE_COMMAND_STOP: 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; - } + case DECODE_COMMAND_SEEK: + return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE; } return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE; diff --git a/src/decoder/_flac_common.h b/src/decoder/_flac_common.h index 68de7e969..2f328afa6 100644 --- a/src/decoder/_flac_common.h +++ b/src/decoder/_flac_common.h @@ -24,132 +24,51 @@ #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> -#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 */ - -/* - * 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 */ - +#include <FLAC/stream_decoder.h> #include <FLAC/metadata.h> -#define FLAC_CHUNK_SIZE 4080 +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "flac" struct flac_data { - unsigned char chunk[FLAC_CHUNK_SIZE]; - float time; - unsigned int bit_rate; - struct audio_format audio_format; - float total_time; + struct pcm_buffer buffer; + + enum sample_format sample_format; + + /** + * The size of one frame in the output buffer. + */ + unsigned frame_size; + + /** + * Is the #stream_info member valid? + */ + bool have_stream_info; + + /** + * A copy of the stream info object passed to the metadata + * callback. Once we drop support for libFLAC 1.1.2, we can + * remove this attribute, and use + * FLAC__stream_decoder_get_total_samples() etc. + */ + FLAC__StreamMetadata_StreamInfo stream_info; + + /** + * 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; + + /** + * The number of the next frame which is going to be decoded. + */ + FLAC__uint64 next_frame; + FLAC__uint64 position; struct decoder *decoder; struct input_stream *input_stream; @@ -162,6 +81,20 @@ void flac_data_init(struct flac_data *data, struct decoder * decoder, struct input_stream *input_stream); +void +flac_data_deinit(struct flac_data *data); + +/** + * Obtains the audio format from the stream_info attribute, and copies + * it to the specified #audio_format object. This also updates the + * frame_size attribute. + * + * @return true on success, false the audio format is not supported + */ +bool +flac_data_get_audio_format(struct flac_data *data, + struct audio_format *audio_format); + void flac_metadata_common_cb(const FLAC__StreamMetadata * block, struct flac_data *data); @@ -169,13 +102,10 @@ 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[]); + const FLAC__int32 *const buf[], + FLAC__uint64 nbytes); #if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7 diff --git a/src/decoder/_ogg_common.c b/src/decoder/_ogg_common.c index 6c6553422..d838e0ff4 100644 --- a/src/decoder/_ogg_common.c +++ b/src/decoder/_ogg_common.c @@ -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..eca5d40e0 100644 --- a/src/decoder/_ogg_common.h +++ b/src/decoder/_ogg_common.h @@ -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_plugin.c index f66d90dc1..18cfdda5d 100644 --- a/src/decoder/audiofile_plugin.c +++ b/src/decoder/audiofile_plugin.c @@ -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); 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; diff --git a/src/decoder/faad_plugin.c b/src/decoder/faad_plugin.c index 7b2806a4c..2272963b9 100644 --- a/src/decoder/faad_plugin.c +++ b/src/decoder/faad_plugin.c @@ -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); } /** @@ -319,7 +330,7 @@ faad_get_file_time_float(const char *file) faacDecConfigurationPtr config; struct input_stream is; - if (!input_stream_open(&is, file)) + if (!input_stream_open(&is, file, NULL)) return -1; buffer = decoder_buffer_new(NULL, &is, @@ -338,8 +349,8 @@ 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); @@ -371,7 +382,7 @@ 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 +419,10 @@ 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 +433,6 @@ 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; @@ -474,15 +478,12 @@ 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, + (size_t)frame_info.samples * 2, bit_rate, NULL); } while (cmd != DECODE_COMMAND_STOP); diff --git a/src/decoder/ffmpeg_plugin.c b/src/decoder/ffmpeg_plugin.c index d2bd642fd..6d856f293 100644 --- a/src/decoder/ffmpeg_plugin.c +++ b/src/decoder/ffmpeg_plugin.c @@ -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> @@ -100,7 +101,7 @@ static int64_t mpd_ffmpeg_seek(URLContext *h, 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; @@ -231,7 +232,6 @@ ffmpeg_send_packet(struct decoder *decoder, struct input_stream *is, 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; @@ -263,22 +263,42 @@ ffmpeg_send_packet(struct decoder *decoder, struct input_stream *is, if (audio_size <= 0) continue; - position = packet->pts != (int64_t)AV_NOPTS_VALUE - ? av_rescale_q(packet->pts, *time_base, - (AVRational){1, 1}) - : 0; + if (packet->pts != (int64_t)AV_NOPTS_VALUE) + decoder_timestamp(decoder, + av_rescale_q(packet->pts, *time_base, + (AVRational){1, 1})); cmd = decoder_data(decoder, is, aligned_buffer, audio_size, - position, codec_context->bit_rate / 1000, NULL); } 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) + int bits = (uint8_t) av_get_bits_per_sample_format(codec_context->sample_fmt); + + /* XXX implement & test other sample formats */ + + switch (bits) { + case 16: + return SAMPLE_FORMAT_S16; + } + + return SAMPLE_FORMAT_UNDEFINED; +#else + /* XXX fixme 16-bit for older ffmpeg (13 Aug 2007) */ + return SAMPLE_FORMAT_S16; +#endif +} + static bool ffmpeg_decode_internal(struct ffmpeg_context *ctx) { + GError *error = NULL; struct decoder *decoder = ctx->decoder; AVCodecContext *codec_context = ctx->codec_context; AVFormatContext *format_context = ctx->format_context; @@ -289,19 +309,12 @@ ffmpeg_decode_internal(struct ffmpeg_context *ctx) 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); + 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); return false; } @@ -357,8 +370,9 @@ static bool ffmpeg_copy_metadata(struct tag *tag, AVMetadata *m, enum tag_type type, const char *name) { - AVMetadataTag *mt = av_metadata_get(m, name, NULL, 0); - if (mt != NULL) + AVMetadataTag *mt = NULL; + + while ((mt = av_metadata_get(m, name, mt, 0)) != NULL) tag_add_item(tag, type, mt->value); return mt != NULL; } @@ -376,35 +390,35 @@ static bool ffmpeg_tag_internal(struct ffmpeg_context *ctx) #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"); - ffmpeg_copy_metadata(tag, f->metadata, TAG_ITEM_ARTIST, "author"); - 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_DATE, "year"); + ffmpeg_copy_metadata(tag, f->metadata, TAG_TITLE, "title"); + ffmpeg_copy_metadata(tag, f->metadata, TAG_ARTIST, "author"); + ffmpeg_copy_metadata(tag, f->metadata, TAG_ALBUM, "album"); + ffmpeg_copy_metadata(tag, f->metadata, TAG_COMMENT, "comment"); + ffmpeg_copy_metadata(tag, f->metadata, TAG_GENRE, "genre"); + ffmpeg_copy_metadata(tag, f->metadata, TAG_TRACK, "track"); + ffmpeg_copy_metadata(tag, f->metadata, TAG_DATE, "year"); #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 @@ -418,7 +432,7 @@ static struct tag *ffmpeg_tag(const char *file) struct ffmpeg_context ctx; bool ret; - if (!input_stream_open(&input, file)) { + if (!input_stream_open(&input, file, NULL)) { g_warning("failed to open %s\n", file); return NULL; } diff --git a/src/decoder/flac_compat.h b/src/decoder/flac_compat.h new file mode 100644 index 000000000..61d2c55e8 --- /dev/null +++ b/src/decoder/flac_compat.h @@ -0,0 +1,114 @@ +/* + * 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. + */ + +/* + * 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_metadata.c b/src/decoder/flac_metadata.c new file mode 100644 index 000000000..1ff99f151 --- /dev/null +++ b/src/decoder/flac_metadata.c @@ -0,0 +1,193 @@ +/* + * 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 "config.h" +#include "flac_metadata.h" +#include "replay_gain.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; +} + +struct replay_gain_info * +flac_parse_replay_gain(const FLAC__StreamMetadata *block) +{ + struct replay_gain_info *rgi; + bool found = false; + + rgi = replay_gain_info_new(); + + found = flac_find_float_comment(block, "replaygain_album_gain", + &rgi->tuples[REPLAY_GAIN_ALBUM].gain) || + flac_find_float_comment(block, "replaygain_album_peak", + &rgi->tuples[REPLAY_GAIN_ALBUM].peak) || + flac_find_float_comment(block, "replaygain_track_gain", + &rgi->tuples[REPLAY_GAIN_TRACK].gain) || + flac_find_float_comment(block, "replaygain_track_peak", + &rgi->tuples[REPLAY_GAIN_TRACK].peak); + if (!found) { + replay_gain_info_free(rgi); + rgi = NULL; + } + + return rgi; +} + +/** + * 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; + } +} diff --git a/src/decoder/flac_metadata.h b/src/decoder/flac_metadata.h new file mode 100644 index 000000000..ef97288d5 --- /dev/null +++ b/src/decoder/flac_metadata.h @@ -0,0 +1,45 @@ +/* + * 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. + */ + +#ifndef MPD_FLAC_METADATA_H +#define MPD_FLAC_METADATA_H + +#include <FLAC/metadata.h> + +struct tag; + +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; +} + +struct replay_gain_info * +flac_parse_replay_gain(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); + +#endif diff --git a/src/decoder/flac_pcm.c b/src/decoder/flac_pcm.c new file mode 100644 index 000000000..a8bf6f293 --- /dev/null +++ b/src/decoder/flac_pcm.c @@ -0,0 +1,108 @@ +/* + * 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 "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_UNDEFINED: + /* unreachable */ + assert(false); + } +} diff --git a/src/decoder/flac_pcm.h b/src/decoder/flac_pcm.h new file mode 100644 index 000000000..4d7a51c4d --- /dev/null +++ b/src/decoder/flac_pcm.h @@ -0,0 +1,33 @@ +/* + * 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. + */ + +#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 index 1e568f70d..ff1d1af3f 100644 --- a/src/decoder/flac_plugin.c +++ b/src/decoder/flac_plugin.c @@ -17,7 +17,14 @@ * 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> @@ -33,8 +40,8 @@ /* this code was based on flac123, from flac-tools */ -static flac_read_status -flac_read_cb(G_GNUC_UNUSED const flac_decoder *fd, +static FLAC__StreamDecoderReadStatus +flac_read_cb(G_GNUC_UNUSED const FLAC__StreamDecoder *fd, FLAC__byte buf[], flac_read_status_size_t *bytes, void *fdata) { @@ -48,53 +55,59 @@ flac_read_cb(G_GNUC_UNUSED const flac_decoder *fd, if (r == 0) { if (decoder_get_command(data->decoder) != DECODE_COMMAND_NONE || input_stream_eof(data->input_stream)) - return flac_read_status_eof; + return FLAC__STREAM_DECODER_READ_STATUS_END_OF_STREAM; else - return flac_read_status_abort; + return FLAC__STREAM_DECODER_READ_STATUS_ABORT; } - return flac_read_status_continue; + return FLAC__STREAM_DECODER_READ_STATUS_CONTINUE; } -static flac_seek_status -flac_seek_cb(G_GNUC_UNUSED const flac_decoder *fd, +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 (!input_stream_seek(data->input_stream, offset, SEEK_SET)) - return flac_seek_status_error; + if (!data->input_stream->seekable) + return FLAC__STREAM_DECODER_SEEK_STATUS_UNSUPPORTED; - return flac_seek_status_ok; + 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_tell_status -flac_tell_cb(G_GNUC_UNUSED const flac_decoder *fd, +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_SEEK_STATUS_UNSUPPORTED; + *offset = (long)(data->input_stream->offset); - return flac_tell_status_ok; + return FLAC__STREAM_DECODER_TELL_STATUS_OK; } -static flac_length_status -flac_length_cb(G_GNUC_UNUSED const flac_decoder *fd, +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_length_status_unsupported; + return FLAC__STREAM_DECODER_LENGTH_STATUS_UNSUPPORTED; *length = (size_t) (data->input_stream->size); - return flac_length_status_ok; + return FLAC__STREAM_DECODER_LENGTH_STATUS_OK; } static FLAC__bool -flac_eof_cb(G_GNUC_UNUSED const flac_decoder *fd, void *fdata) +flac_eof_cb(G_GNUC_UNUSED const FLAC__StreamDecoder *fd, void *fdata) { struct flac_data *data = (struct flac_data *) fdata; @@ -104,7 +117,7 @@ flac_eof_cb(G_GNUC_UNUSED const flac_decoder *fd, void *fdata) } static void -flac_error_cb(G_GNUC_UNUSED const flac_decoder *fd, +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); @@ -143,31 +156,6 @@ static void flacPrintErroredState(FLAC__SeekableStreamDecoderState state) 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) { @@ -199,35 +187,31 @@ static void flacPrintErroredState(FLAC__StreamDecoderState state) } #endif /* FLAC_API_VERSION_CURRENT >= 7 */ -static void flacMetadata(G_GNUC_UNUSED const flac_decoder * dec, +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_decoder *dec, const FLAC__Frame *frame, +flac_write_cb(const FLAC__StreamDecoder *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; + 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); + return flac_common_write(data, frame, buf, nbytes); } static struct tag * @@ -268,12 +252,8 @@ flac_tag_load(const char *file, const char *char_tnum) 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_tag_apply_metadata(tag, char_tnum, block); FLAC__metadata_object_delete(block); } while (FLAC__metadata_simple_iterator_next(it)); @@ -300,6 +280,8 @@ flac_cue_tag_load(const char *file) FLAC__uint64 track_time = 0; #ifdef HAVE_CUE /* libcue */ FLAC__StreamMetadata* vc; + char* cs_filename; + FILE* cs_file; #endif /* libcue */ FLAC__StreamMetadata* si = FLAC__metadata_object_new(FLAC__METADATA_TYPE_STREAMINFO); FLAC__StreamMetadata* cs; @@ -329,16 +311,25 @@ flac_cue_tag_load(const char *file) FLAC__metadata_object_delete(vc); } + + if (tag == NULL) { + cs_filename = g_strconcat(file, ".cue", NULL); + + cs_file = fopen(cs_filename, "rt"); + g_free(cs_filename); + + if (cs_file != NULL) { + tag = cue_tag_file(cs_file, tnum); + fclose(cs_file); + } + } #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); + if (char_tnum != NULL) { + tag_add_item(tag, TAG_TRACK, char_tnum); g_free(char_tnum); } @@ -382,105 +373,167 @@ flac_tag_dup(const char *file) return flac_tag_load(file, NULL); } -static void -flac_decode_internal(struct decoder * decoder, - struct input_stream *input_stream, - bool is_ogg) +/** + * Some glue code around FLAC__stream_decoder_new(). + */ +static FLAC__StreamDecoder * +flac_decoder_new(void) { - 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(); + 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(flac_dec, FLAC__METADATA_TYPE_VORBIS_COMMENT)) - { - g_debug("Failed to set metadata respond\n"); - } + if(!FLAC__stream_decoder_set_metadata_respond(sd, FLAC__METADATA_TYPE_VORBIS_COMMENT)) + g_debug("FLAC__stream_decoder_set_metadata_respond() has failed"); #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; - } - } + return sd; +} - if (!flac_process_metadata(flac_dec)) { - err = "problem reading metadata"; - goto fail; - } +static bool +flac_decoder_initialize(struct flac_data *data, FLAC__StreamDecoder *sd, + bool seekable, FLAC__uint64 duration) +{ + struct audio_format audio_format; - 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; + if (!FLAC__stream_decoder_process_until_end_of_metadata(sd)) { + g_warning("problem reading metadata"); + return false; } - decoder_initialized(decoder, &data.audio_format, - input_stream->seekable, data.total_time); + if (!flac_data_get_audio_format(data, &audio_format)) + return false; + + if (duration == 0) + duration = data->stream_info.total_samples; + + decoder_initialized(data->decoder, &audio_format, + seekable, + (float)duration / + (float)data->stream_info.sample_rate); + return true; +} + +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 (!tag_is_empty(data.tag)) { - cmd = decoder_tag(decoder, input_stream, data.tag); - tag_free(data.tag); - data.tag = tag_new(); + 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 = 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; + FLAC__uint64 seek_sample = t_start + + decoder_seek_where(decoder) * + data->stream_info.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_get_state(flac_dec) == flac_decoder_eof) + FLAC__stream_decoder_get_state(flac_dec) == FLAC__STREAM_DECODER_END_OF_STREAM) break; - if (!flac_process_single(flac_dec)) { + 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_get_state(flac_dec)); - flac_finish(flac_dec); + flacPrintErroredState(FLAC__stream_decoder_get_state(flac_dec)); + FLAC__stream_decoder_finish(flac_dec); } +} -fail: - if (data.replay_gain_info) - replay_gain_info_free(data.replay_gain_info); +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; + } + } - tag_free(data.tag); + if (!flac_decoder_initialize(&data, flac_dec, + input_stream->seekable, 0)) { + flac_data_deinit(&data); + FLAC__stream_decoder_delete(flac_dec); + return; + } - if (flac_dec) - flac_delete(flac_dec); + 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); @@ -509,7 +562,8 @@ flac_container_decode(struct decoder* decoder, FLAC__uint64 track_time = 0; FLAC__StreamMetadata* cs = NULL; - flac_decoder *flac_dec; + FLAC__StreamDecoder *flac_dec; + FLAC__StreamDecoderInitStatus init_status; struct flac_data data; const char *err = NULL; @@ -542,122 +596,43 @@ flac_container_decode(struct decoder* decoder, return; } - if (!(flac_dec = flac_new())) - { - g_free(pathname); + flac_dec = flac_decoder_new(); + if (flac_dec == NULL) 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"; + init_status = is_ogg + ? FLAC__stream_decoder_init_ogg_file(flac_dec, pathname, + flac_write_cb, + flacMetadata, + flac_error_cb, + &data) + : FLAC__stream_decoder_init_file(flac_dec, + pathname, flac_write_cb, + flacMetadata, flac_error_cb, + &data); + g_free(pathname); + if (init_status != FLAC__STREAM_DECODER_INIT_STATUS_OK) { + err = "doing init()"; 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; + if (!flac_decoder_initialize(&data, flac_dec, true, track_time)) { + flac_data_deinit(&data); + FLAC__stream_decoder_delete(flac_dec); + return; } - // 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; + FLAC__stream_decoder_seek_absolute(flac_dec, (FLAC__uint64)t_start); + data.next_frame = t_start; - 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); - } + flac_decoder_loop(&data, flac_dec, t_start, t_end); 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); + flac_data_deinit(&data); + FLAC__stream_decoder_delete(flac_dec); if (err) g_warning("%s\n", err); @@ -672,24 +647,17 @@ flac_filedecode_internal(struct decoder* decoder, const char* fname, bool is_ogg) { - flac_decoder *flac_dec; + FLAC__StreamDecoder *flac_dec; struct flac_data data; const char *err = NULL; unsigned int flac_err_state = 0; - if (!(flac_dec = flac_new())) + flac_dec = flac_decoder_new(); + if (flac_dec == NULL) 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, @@ -727,60 +695,17 @@ flac_filedecode_internal(struct decoder* decoder, } } - 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 (!flac_decoder_initialize(&data, flac_dec, true, 0)) { + flac_data_deinit(&data); + FLAC__stream_decoder_delete(flac_dec); + return; } - if (decoder_get_command(decoder) != DECODE_COMMAND_STOP) - { - flacPrintErroredState(flac_get_state(flac_dec)); - flac_finish(flac_dec); - } + flac_decoder_loop(&data, flac_dec, 0, 0); fail: - if (data.replay_gain_info) - replay_gain_info_free(data.replay_gain_info); - - if (flac_dec) - flac_delete(flac_dec); + flac_data_deinit(&data); + FLAC__stream_decoder_delete(flac_dec); if (err) g_warning("%s\n", err); @@ -836,13 +761,8 @@ oggflac_tag_dup(const char *file) 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; - } + + flac_tag_apply_metadata(ret, NULL, block); } while (FLAC__metadata_iterator_next(it)); FLAC__metadata_iterator_delete(it); @@ -864,7 +784,7 @@ oggflac_decode(struct decoder *decoder, struct input_stream *input_stream) /* 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_decode_internal(decoder, input_stream, true); } diff --git a/src/decoder/fluidsynth_plugin.c b/src/decoder/fluidsynth_plugin.c index 99c874c09..2656b5ebd 100644 --- a/src/decoder/fluidsynth_plugin.c +++ b/src/decoder/fluidsynth_plugin.c @@ -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, NULL); } while (cmd == DECODE_COMMAND_NONE); /* clean up */ diff --git a/src/decoder/mad_plugin.c b/src/decoder/mad_plugin.c index 1ef7183fa..f23ab3c21 100644 --- a/src/decoder/mad_plugin.c +++ b/src/decoder/mad_plugin.c @@ -17,10 +17,11 @@ * 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 "audio_check.h" #include <assert.h> #include <unistd.h> @@ -164,7 +165,7 @@ 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); @@ -779,10 +780,10 @@ 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; @@ -792,7 +793,7 @@ 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); @@ -804,7 +805,7 @@ 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); @@ -919,7 +920,7 @@ static int mp3_total_file_time(const char *file) struct mp3_data data; int ret; - if (!input_stream_open(&input_stream, file)) + if (!input_stream_open(&input_stream, file, NULL)) return -1; mp3_data_init(&data, NULL, &input_stream); if (!mp3_decode_first_frame(&data, NULL, NULL)) @@ -1024,7 +1025,6 @@ 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); if (cmd != DECODE_COMMAND_NONE) @@ -1170,17 +1170,11 @@ 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; @@ -1192,7 +1186,21 @@ mp3_decode(struct decoder *decoder, struct input_stream *input_stream) 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); + if (replay_gain_info != NULL) + replay_gain_info_free(replay_gain_info); + mp3_data_finish(&data); + return; + } decoder_initialized(decoder, &audio_format, data.input_stream->seekable, data.total_time); diff --git a/src/decoder/mikmod_plugin.c b/src/decoder/mikmod_plugin.c index 065c34319..bbb330cb7 100644 --- a/src/decoder/mikmod_plugin.c +++ b/src/decoder/mikmod_plugin.c @@ -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 <mikmod.h> +#include <assert.h> #undef G_LOG_DOMAIN #define G_LOG_DOMAIN "mikmod" @@ -29,30 +31,34 @@ #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 +74,18 @@ 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 +100,19 @@ 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)) + g_error("Invalid sample rate in line %d: %u", + param->line, mikmod_sample_rate); + md_device = 0; md_reverb = 0; @@ -106,7 +120,7 @@ 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); @@ -119,115 +133,83 @@ 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, NULL); } - 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; + MODULE *handle; char *title; - path2 = g_strdup(file); - moduleHandle = Player_Load(path2, 128, 0); + path2 = g_strdup(path_fs); + handle = Player_Load(path2, 128, 0); g_free(path2); - if (moduleHandle == NULL) { - g_debug("Failed to open file: %s", file); + if (handle == NULL) { + g_debug("Failed to open file: %s", path_fs); return NULL; } - Player_Free(moduleHandle); + Player_Free(handle); ret = tag_new(); ret->time = 0; - path2 = g_strdup(file); + path2 = g_strdup(path_fs); title = g_strdup(Player_LoadTitle(path2)); g_free(path2); if (title) - tag_add_item(ret, TAG_ITEM_TITLE, title); + tag_add_item(ret, TAG_TITLE, title); return ret; } -static const char *const modSuffixes[] = { +static const char *const mikmod_decoder_suffixes[] = { "amf", "dsm", "far", @@ -248,9 +230,9 @@ 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_plugin.c index f636f2fa6..05c9ef2d7 100644 --- a/src/decoder/modplug_plugin.c +++ b/src/decoder/modplug_plugin.c @@ -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, NULL); 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); } @@ -168,7 +157,7 @@ static struct tag *mod_tagdup(const char *file) char *title; struct input_stream is; - if (!input_stream_open(&is, file)) { + if (!input_stream_open(&is, file, NULL)) { g_warning("cant open file %s\n", file); return NULL; } @@ -186,11 +175,11 @@ static struct tag *mod_tagdup(const char *file) 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); diff --git a/src/decoder/mp4ff_plugin.c b/src/decoder/mp4ff_plugin.c index cf9382904..325a97291 100644 --- a/src/decoder/mp4ff_plugin.c +++ b/src/decoder/mp4ff_plugin.c @@ -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> @@ -98,7 +99,7 @@ mp4_seek(void *user_data, uint64_t position) { struct mp4_context *ctx = user_data; - return input_stream_seek(ctx->input_stream, position, SEEK_SET) + return input_stream_seek(ctx->input_stream, position, SEEK_SET, NULL) ? 0 : -1; } @@ -110,6 +111,7 @@ 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(); @@ -130,22 +132,17 @@ 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; } @@ -265,7 +262,7 @@ 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) { @@ -331,7 +328,7 @@ 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, NULL); } g_free(seek_table); @@ -359,7 +356,7 @@ mp4_tag_dup(const char *file) int32_t scale; int i; - if (!input_stream_open(&input_stream, file)) { + if (!input_stream_open(&input_stream, file, NULL)) { g_warning("Failed to open file: %s", file); return NULL; } @@ -395,22 +392,22 @@ mp4_tag_dup(const char *file) mp4ff_meta_get_by_index(mp4fh, i, &item, &value); if (0 == g_ascii_strcasecmp("artist", item)) { - tag_add_item(ret, TAG_ITEM_ARTIST, value); + tag_add_item(ret, TAG_ARTIST, value); } else if (0 == g_ascii_strcasecmp("title", item)) { - tag_add_item(ret, TAG_ITEM_TITLE, value); + tag_add_item(ret, TAG_TITLE, value); } else if (0 == g_ascii_strcasecmp("album", item)) { - tag_add_item(ret, TAG_ITEM_ALBUM, value); + tag_add_item(ret, TAG_ALBUM, value); } else if (0 == g_ascii_strcasecmp("track", item)) { - tag_add_item(ret, TAG_ITEM_TRACK, value); + tag_add_item(ret, TAG_TRACK, value); } else if (0 == g_ascii_strcasecmp("disc", item)) { /* Is that the correct id? */ - tag_add_item(ret, TAG_ITEM_DISC, value); + tag_add_item(ret, TAG_DISC, value); } else if (0 == g_ascii_strcasecmp("genre", item)) { - tag_add_item(ret, TAG_ITEM_GENRE, value); + tag_add_item(ret, TAG_GENRE, value); } else if (0 == g_ascii_strcasecmp("date", item)) { - tag_add_item(ret, TAG_ITEM_DATE, value); + tag_add_item(ret, TAG_DATE, value); } else if (0 == g_ascii_strcasecmp("writer", item)) { - tag_add_item(ret, TAG_ITEM_COMPOSER, value); + tag_add_item(ret, TAG_COMPOSER, value); } free(item); diff --git a/src/decoder/mpcdec_plugin.c b/src/decoder/mpcdec_plugin.c index 26349f93a..a186bc368 100644 --- a/src/decoder/mpcdec_plugin.c +++ b/src/decoder/mpcdec_plugin.c @@ -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> @@ -27,6 +28,7 @@ #endif #include <glib.h> +#include <assert.h> #include <unistd.h> #undef G_LOG_DOMAIN @@ -58,7 +60,7 @@ 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 @@ -140,6 +142,7 @@ 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; @@ -149,10 +152,8 @@ 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; @@ -193,18 +194,14 @@ 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; } @@ -220,16 +217,14 @@ mpcdec_decode(struct decoder *mpd_decoder, struct input_stream *is) 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) @@ -260,19 +255,15 @@ 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); } while (cmd != DECODE_COMMAND_STOP); @@ -296,7 +287,7 @@ mpcdec_get_file_duration(const char *file) mpc_streaminfo info; struct mpc_decoder_data data; - if (!input_stream_open(&is, file)) + if (!input_stream_open(&is, file, NULL)) return -1; data.is = &is; diff --git a/src/decoder/mpg123_decoder_plugin.c b/src/decoder/mpg123_decoder_plugin.c new file mode 100644 index 000000000..0d7cb9e0e --- /dev/null +++ b/src/decoder/mpg123_decoder_plugin.c @@ -0,0 +1,210 @@ +/* + * 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 "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, NULL); + + /* 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_plugin.c index bdd589ccb..b8b087b47 100644 --- a/src/decoder/oggflac_plugin.c +++ b/src/decoder/oggflac_plugin.c @@ -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 */ @@ -261,12 +245,21 @@ fail: static struct tag * oggflac_tag_dup(const char *file) { + GError *error = NULL; struct input_stream input_stream; OggFLAC__SeekableStreamDecoder *decoder; struct flac_data data; + struct tag *tag; + + if (!input_stream_open(&input_stream, file, &error)) { + if (error != NULL) { + g_warning("%s", error->message); + g_error_free(error); + } - if (!input_stream_open(&input_stream, file)) return NULL; + } + if (ogg_stream_type_detect(&input_stream) != FLAC) { input_stream_close(&input_stream); return NULL; @@ -274,7 +267,7 @@ oggflac_tag_dup(const char *file) /* 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, NULL, &input_stream); @@ -284,15 +277,18 @@ 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); + oggflac_cleanup(decoder); input_stream_close(&input_stream); - 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 +296,14 @@ 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 +311,13 @@ 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 (!flac_data_get_audio_format(&data, &audio_format)) 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.stream_info.total_samples / + (float)data.stream_info.sample_rate); while (true) { OggFLAC__seekable_stream_decoder_process_single(decoder); @@ -333,11 +327,10 @@ 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.stream_info.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 +345,8 @@ 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 }; diff --git a/src/decoder/sidplay_plugin.cxx b/src/decoder/sidplay_plugin.cxx index c62e6b4b6..9bfe98f3d 100644 --- a/src/decoder/sidplay_plugin.cxx +++ b/src/decoder/sidplay_plugin.cxx @@ -17,18 +17,186 @@ * 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) { @@ -36,13 +204,19 @@ sidplay_file_decode(struct decoder *decoder, const char *path_fs) /* load the tune */ - SidTune tune(path_fs, NULL, true); + 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; } - tune.selectSong(1); + 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 */ @@ -67,7 +241,7 @@ sidplay_file_decode(struct decoder *decoder, const char *path_fs) return; } - builder.filter(false); + builder.filter(filter_setting); if (!builder) { g_warning("ReSIDBuilder.filter() failed"); return; @@ -103,14 +277,16 @@ sidplay_file_decode(struct decoder *decoder, const char *path_fs) /* initialize the MPD decoder */ struct audio_format audio_format; - audio_format.sample_rate = 48000; - audio_format.bits = 16; - audio_format.channels = 2; + audio_format_init(&audio_format, 48000, SAMPLE_FORMAT_S16, 2); + assert(audio_format_valid(&audio_format)); - decoder_initialized(decoder, &audio_format, false, -1); + 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]; @@ -120,30 +296,108 @@ sidplay_file_decode(struct decoder *decoder, const char *path_fs) if (nbytes == 0) break; + decoder_timestamp(decoder, (double)player.time() / timebase); + cmd = decoder_data(decoder, NULL, buffer, nbytes, - 0, 0, NULL); - } while (cmd == DECODE_COMMAND_NONE); + 0, NULL); + + 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) { - SidTune tune(path_fs, NULL, true); + 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) - tag_add_item(tag, TAG_ITEM_TITLE, info.infoString[0]); - + 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_ITEM_ARTIST, info.infoString[1]); + 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", NULL @@ -152,12 +406,12 @@ static const char *const sidplay_suffixes[] = { extern const struct decoder_plugin sidplay_decoder_plugin; const struct decoder_plugin sidplay_decoder_plugin = { "sidplay", - NULL, /* init() */ - NULL, /* finish() */ + sidplay_init, + sidplay_finish, NULL, /* stream_decode() */ sidplay_file_decode, sidplay_tag_dup, - NULL, /* container_scan */ + sidplay_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..fd51f4a33 --- /dev/null +++ b/src/decoder/sndfile_decoder_plugin.c @@ -0,0 +1,251 @@ +/* + * 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 "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, NULL); + if (cmd == DECODE_COMMAND_SEEK) { + sf_count_t c = + time_to_frame(decoder_seek_where(decoder), + &audio_format); + c = sf_seek(sf, c, SEEK_SET); + if (c < 0) + decoder_seek_error(decoder); + else + decoder_command_finished(decoder); + cmd = DECODE_COMMAND_NONE; + } + } while (cmd == DECODE_COMMAND_NONE); + + sf_close(sf); +} + +static struct tag * +sndfile_tag_dup(const char *path_fs) +{ + SNDFILE *sf; + SF_INFO info; + struct tag *tag; + const char *p; + + info.format = 0; + + sf = sf_open(path_fs, SFM_READ, &info); + if (sf == NULL) + return NULL; + + if (!audio_valid_sample_rate(info.samplerate)) { + sf_close(sf); + g_warning("Invalid sample rate in %s\n", path_fs); + return NULL; + } + + tag = tag_new(); + tag->time = info.frames / info.samplerate; + + p = sf_get_string(sf, SF_STR_TITLE); + if (p != NULL) + tag_add_item(tag, TAG_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_plugin.c index 0ff898647..b323cf775 100644..100755 --- a/src/decoder/vorbis_plugin.c +++ b/src/decoder/vorbis_plugin.c @@ -17,13 +17,13 @@ * 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> @@ -55,17 +55,17 @@ #define OGG_DECODE_USE_BIGENDIAN 0 #endif -typedef struct _OggCallbackData { +struct vorbis_decoder_data { 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) { + struct vorbis_decoder_data *data = (struct vorbis_decoder_data *)vdata; size_t ret; - OggCallbackData *data = (OggCallbackData *) vdata; ret = decoder_read(data->decoder, data->input_stream, ptr, size * nmemb); @@ -76,11 +76,11 @@ static size_t ogg_read_cb(void *ptr, size_t size, size_t nmemb, void *vdata) static int ogg_seek_cb(void *vdata, ogg_int64_t offset, int whence) { - const OggCallbackData *data = (const OggCallbackData *) vdata; + struct vorbis_decoder_data *data = (struct vorbis_decoder_data *)vdata; return data->seekable && decoder_get_command(data->decoder) != DECODE_COMMAND_STOP && - input_stream_seek(data->input_stream, offset, whence) + input_stream_seek(data->input_stream, offset, whence, NULL) ? 0 : -1; } @@ -92,12 +92,37 @@ static int ogg_close_cb(G_GNUC_UNUSED void *vdata) static long ogg_tell_cb(void *vdata) { - const OggCallbackData *data = (const OggCallbackData *) vdata; + const struct vorbis_decoder_data *data = + (const struct vorbis_decoder_data *)vdata; return (long)data->input_stream->offset; } 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 const char * vorbis_comment_value(const char *comment, const char *needle) { size_t len = strlen(needle); @@ -176,11 +201,11 @@ 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) @@ -228,7 +253,7 @@ oggvorbis_seekable(struct decoder *decoder) uri = decoder_get_uri(decoder); /* 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 */ + delays song playback by 10 or 20 seconds */ seekable = !uri_has_scheme(uri); g_free(uri); @@ -240,10 +265,12 @@ static void vorbis_stream_decode(struct decoder *decoder, struct input_stream *input_stream) { + GError *error = NULL; OggVorbis_File vf; ov_callbacks callbacks; - OggCallbackData data; + struct vorbis_decoder_data data; struct audio_format audio_format; + float total_time; int current_section; int prev_section = -1; long ret; @@ -251,8 +278,7 @@ vorbis_stream_decode(struct decoder *decoder, 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) @@ -260,7 +286,7 @@ 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); + input_stream_seek(input_stream, 0, SEEK_SET, NULL); data.decoder = decoder; data.input_stream = input_stream; @@ -271,36 +297,33 @@ vorbis_stream_decode(struct decoder *decoder, callbacks.close_func = ogg_close_cb; callbacks.tell_func = ogg_tell_cb; if ((ret = ov_open_callbacks(&data, &vf, NULL, 0, callbacks)) < 0) { - const char *error; - if (decoder_get_command(decoder) != DECODE_COMMAND_NONE) 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; - } + g_warning("Error decoding Ogg Vorbis stream: %s", + vorbis_strerror(ret)); + return; + } - g_warning("Error decoding Ogg Vorbis stream: %s", error); + vi = ov_info(&vf, -1); + if (vi == NULL) { + g_warning("ov_info() has failed"); return; } - audio_format.bits = 16; + + 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; + } + + total_time = ov_time_total(&vf, -1); + if (total_time < 0) + total_time = 0; + + decoder_initialized(decoder, &audio_format, data.seekable, total_time); do { if (cmd == DECODE_COMMAND_SEEK) { @@ -320,30 +343,23 @@ vorbis_stream_decode(struct decoder *decoder, break; if (current_section != prev_section) { - /*printf("new song!\n"); */ - vorbis_info *vi = ov_info(&vf, -1); + char **comments; 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); + 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); @@ -352,16 +368,15 @@ vorbis_stream_decode(struct decoder *decoder, replay_gain_info_free(replay_gain_info); replay_gain_info = new_rgi; } - } - prev_section = current_section; + 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); } while (cmd != DECODE_COMMAND_STOP); @@ -378,7 +393,7 @@ vorbis_tag_dup(const char *file) FILE *fp; OggVorbis_File vf; - fp = fopen(file, "r"); + fp = fopen(file, "rb"); if (!fp) { return NULL; } diff --git a/src/decoder/wavpack_plugin.c b/src/decoder/wavpack_plugin.c index 7ad3a62b0..fbcebb803 100644 --- a/src/decoder/wavpack_plugin.c +++ b/src/decoder/wavpack_plugin.c @@ -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; - } } } @@ -129,6 +123,33 @@ 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. @@ -137,30 +158,27 @@ static void wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek, struct replay_gain_info *replay_gain_info) { + 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 +198,6 @@ 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 +205,6 @@ 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 +224,6 @@ 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, @@ -221,7 +233,7 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek, decoder_data( decoder, NULL, chunk, samples_got * output_sample_size, - current_time, bitrate, + bitrate, replay_gain_info ); } @@ -397,13 +409,13 @@ 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 @@ -474,7 +486,7 @@ wavpack_open_wvc(struct decoder *decoder, struct input_stream *is_wvc, wvc_url = g_strconcat(utf8url, "c", NULL); g_free(utf8url); - ret = input_stream_open(is_wvc, wvc_url); + ret = input_stream_open(is_wvc, wvc_url, NULL); g_free(wvc_url); if (!ret) { @@ -508,7 +520,7 @@ 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; + int open_flags = OPEN_NORMALIZE; struct wavpack_input isp, isp_wvc; bool can_seek = is->seekable; @@ -553,7 +565,7 @@ wavpack_filedecode(struct decoder *decoder, const char *fname) 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( diff --git a/src/decoder/wildmidi_plugin.c b/src/decoder/wildmidi_plugin.c index 8bad6943a..0392c06b9 100644 --- a/src/decoder/wildmidi_plugin.c +++ b/src/decoder/wildmidi_plugin.c @@ -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; @@ -91,8 +92,6 @@ wildmidi_file_decode(struct decoder *decoder, const char *path_fs) break; cmd = decoder_data(decoder, NULL, buffer, len, - (float)info->current_sample / - (float)WILDMIDI_SAMPLE_RATE, 0, NULL); if (cmd == DECODE_COMMAND_SEEK) { |