/* * Copyright (C) 2003-2010 The Music Player Daemon Project * http://www.musicpd.org * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * You should have received a copy of the GNU General Public License along * with this program; if not, write to the Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ #include "config.h" #include "decoder_api.h" #include "audio_check.h" #include <audiofile.h> #include <af_vfs.h> #include <assert.h> #include <glib.h> #include <stdio.h> #undef G_LOG_DOMAIN #define G_LOG_DOMAIN "audiofile" /* pick 1020 since its devisible for 8,16,24, and 32-bit audio */ #define CHUNK_SIZE 1020 static int audiofile_get_duration(const char *file) { int total_time; AFfilehandle af_fp = afOpenFile(file, "r", NULL); if (af_fp == AF_NULL_FILEHANDLE) { return -1; } total_time = (int) ((double)afGetFrameCount(af_fp, AF_DEFAULT_TRACK) / afGetRate(af_fp, AF_DEFAULT_TRACK)); afCloseFile(af_fp); return total_time; } static ssize_t audiofile_file_read(AFvirtualfile *vfile, void *data, size_t length) { struct input_stream *is = (struct input_stream *) vfile->closure; 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 audiofile_file_length(AFvirtualfile *vfile) { struct input_stream *is = (struct input_stream *) vfile->closure; return is->size; } static long audiofile_file_tell(AFvirtualfile *vfile) { struct input_stream *is = (struct input_stream *) vfile->closure; return is->offset; } static void audiofile_file_destroy(AFvirtualfile *vfile) { assert(vfile->closure != NULL); vfile->closure = NULL; } static long 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, NULL)) { return is->offset; } else { return -1; } } static AFvirtualfile * setup_virtual_fops(struct input_stream *stream) { AFvirtualfile *vf = g_malloc(sizeof(AFvirtualfile)); vf->closure = stream; vf->write = NULL; vf->read = audiofile_file_read; vf->length = audiofile_file_length; vf->destroy = audiofile_file_destroy; vf->seek = audiofile_file_seek; vf->tell = audiofile_file_tell; 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; struct audio_format audio_format; float total_time; uint16_t bit_rate; int ret; char chunk[CHUNK_SIZE]; enum decoder_command cmd; if (!is->seekable) { g_warning("not seekable"); return; } vf = setup_virtual_fops(is); af_fp = afOpenVirtualFile(vf, "r", NULL); if (af_fp == AF_NULL_FILEHANDLE) { g_warning("failed to input stream\n"); return; } 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; } frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK); total_time = ((float)frame_count / (float)audio_format.sample_rate); bit_rate = (uint16_t)(is->size * 8.0 / total_time / 1000.0 + 0.5); fs = (int)afGetVirtualFrameSize(af_fp, AF_DEFAULT_TRACK, 1); decoder_initialized(decoder, &audio_format, true, total_time); do { ret = afReadFrames(af_fp, AF_DEFAULT_TRACK, chunk, CHUNK_SIZE / fs); if (ret <= 0) break; cmd = decoder_data(decoder, NULL, chunk, ret * fs, bit_rate); if (cmd == DECODE_COMMAND_SEEK) { AFframecount frame = decoder_seek_where(decoder) * audio_format.sample_rate; afSeekFrame(af_fp, AF_DEFAULT_TRACK, frame); decoder_command_finished(decoder); cmd = DECODE_COMMAND_NONE; } } while (cmd == DECODE_COMMAND_NONE); afCloseFile(af_fp); } static struct tag *audiofile_tag_dup(const char *file) { struct tag *ret = NULL; int total_time = audiofile_get_duration(file); if (total_time >= 0) { ret = tag_new(); ret->time = total_time; } else { g_debug("Failed to get total song time from: %s\n", file); } return ret; } static const char *const audiofile_suffixes[] = { "wav", "au", "aiff", "aif", NULL }; static const char *const audiofile_mime_types[] = { "audio/x-wav", "audio/x-aiff", NULL }; const struct decoder_plugin audiofile_decoder_plugin = { .name = "audiofile", .stream_decode = audiofile_stream_decode, .tag_dup = audiofile_tag_dup, .suffixes = audiofile_suffixes, .mime_types = audiofile_mime_types, };