From 9b9abff97272b52f133ff23addd58b6a90a49a73 Mon Sep 17 00:00:00 2001 From: Max Kellermann Date: Mon, 4 Jan 2010 21:36:33 +0100 Subject: renamed decoder plugin sources Make it X_decoder_plugin.c. --- src/decoder/audiofile_decoder_plugin.c | 256 +++++++++++++++++++++++++++++++++ 1 file changed, 256 insertions(+) create mode 100644 src/decoder/audiofile_decoder_plugin.c (limited to 'src/decoder/audiofile_decoder_plugin.c') diff --git a/src/decoder/audiofile_decoder_plugin.c b/src/decoder/audiofile_decoder_plugin.c new file mode 100644 index 000000000..3026f3cc7 --- /dev/null +++ b/src/decoder/audiofile_decoder_plugin.c @@ -0,0 +1,256 @@ +/* + * 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 +#include +#include +#include + +#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, +}; -- cgit v1.2.3