Mercurial > audlegacy-plugins
view src/flacng/plugin.c @ 2204:43da009d6e76
add "fla" to the extension list. fix #6.
author | Yoshiki Yazawa <yaz@cc.rim.or.jp> |
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date | Mon, 03 Dec 2007 18:50:19 +0900 |
parents | e5401b6c4665 |
children | 5395c85a8724 |
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/* * A FLAC decoder plugin for the Audacious Media Player * Copyright (C) 2005 Ralf Ertzinger * * 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., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. */ #include "flacng.h" #include <audacious/util.h> #include <audacious/output.h> #include <audacious/i18n.h> #include "tools.h" #include "plugin.h" #include "seekable_stream_callbacks.h" #include "version.h" #include "debug.h" static gchar *flac_fmts[] = { "flac", "fla", NULL }; InputPlugin flac_ip = { .description = "FLACng Audio Plugin", .init = flac_init, .about = flac_aboutbox, .is_our_file = flac_is_our_file, .play_file = flac_play_file, .stop = flac_stop, .pause = flac_pause, .seek = flac_seek, .get_song_info = flac_get_song_info, .get_song_tuple = flac_get_song_tuple, // get a tuple .is_our_file_from_vfs = flac_is_our_fd, // version of is_our_file which is handed an FD .vfs_extensions = flac_fmts // vector of fileextensions allowed by the plugin }; InputPlugin *flac_iplist[] = { &flac_ip, NULL }; DECLARE_PLUGIN(flacng, NULL, NULL, flac_iplist, NULL, NULL, NULL, NULL, NULL); FLAC__StreamDecoder* test_decoder; FLAC__StreamDecoder* main_decoder; callback_info* test_info; callback_info* main_info; gboolean plugin_initialized = FALSE; gint seek_to = -1; static GThread* thread = NULL; /* === */ void flac_init(void) { FLAC__StreamDecoderInitStatus ret; _ENTER; /* * Callback structure and decoder for file test * purposes */ if (NULL == (test_info = init_callback_info("test"))) { _ERROR("Could not initialize the test callback structure!"); _LEAVE; } _DEBUG("Test callback structure at %p", test_info); if (NULL == (test_decoder = FLAC__stream_decoder_new())) { _ERROR("Could not create the test FLAC decoder instance!"); _LEAVE; } /* * We want the VORBISCOMMENT metadata, for the file tags * and SEEKTABLE */ FLAC__stream_decoder_set_metadata_respond(test_decoder, FLAC__METADATA_TYPE_VORBIS_COMMENT); FLAC__stream_decoder_set_metadata_respond(test_decoder, FLAC__METADATA_TYPE_SEEKTABLE); /* * Callback structure and decoder for main decoding * loop */ if (NULL == (main_info = init_callback_info("main"))) { _ERROR("Could not initialize the main callback structure!"); _LEAVE; } _DEBUG("Main callback structure at %p", main_info); if (NULL == (main_decoder = FLAC__stream_decoder_new())) { _ERROR("Could not create the main FLAC decoder instance!"); _LEAVE; } /* * We want the VORBISCOMMENT metadata, for the file tags * and SEEKTABLE */ FLAC__stream_decoder_set_metadata_respond(main_decoder, FLAC__METADATA_TYPE_VORBIS_COMMENT); FLAC__stream_decoder_set_metadata_respond(main_decoder, FLAC__METADATA_TYPE_SEEKTABLE); /* * Initialize decoders */ if (FLAC__STREAM_DECODER_INIT_STATUS_OK != (ret = FLAC__stream_decoder_init_stream( test_decoder, read_callback, seek_callback, tell_callback, length_callback, eof_callback, write_callback, metadata_callback, error_callback, test_info))) { _ERROR("Could not initialize test FLAC decoder: %s(%d)", FLAC__StreamDecoderInitStatusString[ret], ret); _LEAVE; } if (FLAC__STREAM_DECODER_INIT_STATUS_OK != (ret = FLAC__stream_decoder_init_stream( main_decoder, read_callback, seek_callback, tell_callback, length_callback, eof_callback, write_callback, metadata_callback, error_callback, main_info))) { _ERROR("Could not initialize main FLAC decoder: %s(%d)", FLAC__StreamDecoderInitStatusString[ret], ret); _LEAVE; } _DEBUG("plugin initialized OK!"); plugin_initialized = TRUE; _LEAVE; } /* --- */ gboolean flac_is_our_fd(gchar* filename, VFSFile* fd) { _ENTER; if (!plugin_initialized) { _ERROR("Plugin not initialized!"); _LEAVE FALSE; } _DEBUG("Testing fd for file: %s", filename); INFO_LOCK(test_info); if (FALSE == read_metadata(fd, test_decoder, test_info)) { _DEBUG("File not handled by this plugin!"); INFO_UNLOCK(test_info); _LEAVE FALSE; } /* * See if the metadata has changed */ if (FALSE == test_info->metadata_changed) { _DEBUG("No metadata found in stream"); INFO_UNLOCK(test_info); _LEAVE FALSE; } /* * If we get here, the file is supported by FLAC. * The stream characteristics have been filled in by * the metadata callback. */ _DEBUG("Stream encoded at %d Hz, %d bps, %d channels", test_info->stream.samplerate, test_info->stream.bits_per_sample, test_info->stream.channels); if (MAX_SUPPORTED_CHANNELS < test_info->stream.channels) { _ERROR("This number of channels (%d) is currently not supported, stream not handled by this plugin", test_info->stream.channels); INFO_UNLOCK(test_info); _LEAVE FALSE; } if ((16 != test_info->stream.bits_per_sample) && (24 != test_info->stream.bits_per_sample) && (8 != test_info->stream.bits_per_sample)) { _ERROR("This number of bits (%d) is currently not supported, stream not handled by this plugin", test_info->stream.bits_per_sample); INFO_UNLOCK(test_info); _LEAVE FALSE; } /* * Looks good. */ _DEBUG("Accepting file %s", filename); reset_info(test_info, FALSE); INFO_UNLOCK(test_info); _LEAVE TRUE; } /* --- */ gboolean flac_is_our_file(gchar* filename) { VFSFile* fd; gboolean ret; _ENTER; _DEBUG("Testing file: %s", filename); /* * Open the file */ if (NULL == (fd = aud_vfs_fopen(filename, "rb"))) { _ERROR("Could not open file for reading! (%s)", filename); _LEAVE FALSE; } ret = flac_is_our_fd(filename, fd); aud_vfs_fclose(fd); _LEAVE ret; } /* --- */ void squeeze_audio(gint32* src, void* dst, guint count, guint src_res, guint dst_res) { /* * Changes the sample width of count samples in src from * src_res to dst_res and places the result in dst */ gint i; gint32* rp = src; gint8* wp = dst; gint16* wp2 = dst; gint32* wp4 = dst; _ENTER; _DEBUG("Converting %d samples %d->%d", count, src_res, dst_res); if ((src_res % 8 != 0) || (dst_res % 8 != 0)) { _ERROR("Can not convert from %d bps to %d bps: not a multiple of 8", src_res, dst_res); _LEAVE; } if (16 == dst_res) { if (8 == src_res) { for (i=0; i<count; i++, wp2++, rp++) { *wp2 = (*rp << 8) & 0xffff; } } else if (16 == src_res) { for (i=0; i<count; i++, wp2++, rp++) { *wp2 = *rp & 0xffff; } } else if (24 == src_res) { for (i=0; i<count; i++, wp2++, rp++) { *wp2 = (*rp >> 8) & 0xffff; } } else if (32 == src_res) { for (i=0; i<count; i++, wp2++, rp++) { *wp2 = (*rp >> 16) & 0xffff; } } } else if (8 == dst_res) { if (8 == src_res) { for (i=0; i<count; i++, wp++, rp++) { *wp = *rp & 0xff; } } else if (16 == src_res) { for (i=0; i<count; i++, wp++, rp++) { *wp = (*rp >> 8) & 0xff; } } else if (24 == src_res) { for (i=0; i<count; i++, wp++, rp++) { *wp = (*rp >> 16) & 0xff; } } else if (32 == src_res) { for (i=0; i<count; i++, wp++, rp++) { *wp = (*rp >> 24) & 0xff; } } } else if (32 == dst_res) { if (8 == src_res) { for (i=0; i<count; i++, wp4++, rp++) { *wp4 = (*rp << 24) & 0xffffffff; } } else if (16 == src_res) { for (i=0; i<count; i++, wp4++, rp++) { *wp4 = (*rp << 16) & 0xffffffff; } } else if (24 == src_res) { for (i=0; i<count; i++, wp4++, rp++) { *wp4 = (*rp << 8) & 0xffffffff; } } else if (32 == src_res) { for (i=0; i<count; i++, wp4++, rp++) { *wp4 = *rp; } } } _LEAVE; } /* --- */ static gpointer flac_play_loop(gpointer arg) { /* * The main play loop. * Decode a frame, push the decoded data to the output plugin * chunkwise. Repeat until finished. */ gint32* read_pointer; gint elements_left; gint seek_sample; FLAC__StreamDecoderState state; struct stream_info stream_info; guint sample_count; void* play_buffer; InputPlayback* playback = (InputPlayback *) arg; _ENTER; if (NULL == (play_buffer = malloc(BUFFER_SIZE_BYTE))) { _ERROR("Could not allocate conversion buffer"); playback->playing = FALSE; _LEAVE NULL; } stream_info.samplerate = main_info->stream.samplerate; stream_info.channels = main_info->stream.channels; main_info->metadata_changed = FALSE; if (!playback->output->open_audio(FMT_S16_NE, main_info->stream.samplerate, main_info->stream.channels)) { playback->playing = FALSE; _ERROR("Could not open output plugin!"); _LEAVE NULL; } while (TRUE == playback->playing) { /* * Try to decode a single frame of audio */ if (FALSE == FLAC__stream_decoder_process_single(main_decoder)) { /* * Something went wrong */ _ERROR("Error while decoding!"); break; } /* * Maybe the metadata changed midstream. Check. */ if (main_info->metadata_changed) { /* * Samplerate and channels are important */ if (stream_info.samplerate != main_info->stream.samplerate) { _ERROR("Samplerate changed midstream (now: %d, was: %d). This is not supported yet.", main_info->stream.samplerate, stream_info.samplerate); break; } if (stream_info.channels != main_info->stream.channels) { _ERROR("Number of channels changed midstream (now: %d, was: %d). This is not supported yet.", main_info->stream.channels, stream_info.channels); break; } main_info->metadata_changed = FALSE; } /* * Compare the frame metadata to the current stream metadata */ if (main_info->stream.samplerate != main_info->frame.samplerate) { _ERROR("Frame samplerate mismatch (stream: %d, frame: %d)!", main_info->stream.samplerate, main_info->frame.samplerate); break; } if (main_info->stream.channels != main_info->frame.channels) { _ERROR("Frame channel mismatch (stream: %d, frame: %d)!", main_info->stream.channels, main_info->frame.channels); break; } /* * If the frame decoded was an audio frame we now * have data in info->output_buffer * * The data in this buffer is in 32 bit wide samples, even if the * real sample width is smaller. It has to be converted before * feeding it to he output plugin. * * Feed the data in chunks of OUTPUT_BLOCK_SIZE elements to the output * plugin */ read_pointer = main_info->output_buffer; elements_left = main_info->buffer_used; while ((TRUE == playback->playing) && (0 != elements_left)) { sample_count = MIN(OUTPUT_BLOCK_SIZE, elements_left); /* * Convert the audio. * Currently this is hardcoded to 16 bit. * 8 and 24 bit are sampled up/down linear. */ _DEBUG("Converting %d samples to FMT_S16_NE", sample_count); squeeze_audio(read_pointer, play_buffer, sample_count, main_info->frame.bits_per_sample, 16); _DEBUG("Copying %d samples to output plugin", sample_count); playback->pass_audio(playback, FMT_S16_NE, main_info->frame.channels, sample_count * sizeof(gint16), play_buffer, NULL); read_pointer += sample_count; elements_left -= sample_count; _DEBUG("%d elements left to be output", elements_left); } /* * Clear the buffer. */ main_info->write_pointer = main_info->output_buffer; main_info->buffer_free = BUFFER_SIZE_SAMP; main_info->buffer_used = 0; /* * Do we have to seek to somewhere? */ if (-1 != seek_to) { _DEBUG("Seek requested to %d seconds", seek_to); seek_sample = seek_to * main_info->stream.samplerate; _DEBUG("Seek requested to sample %d", seek_sample); if (FALSE == FLAC__stream_decoder_seek_absolute(main_decoder, seek_sample)) { _ERROR("Could not seek to sample %d!", seek_sample); } else { /* * Flush the buffers */ flac_ip.output->flush(seek_to * 1000); } seek_to = -1; } /* * Have we reached the end of the stream? */ state = FLAC__stream_decoder_get_state(main_decoder); if (0 == elements_left && (FLAC__STREAM_DECODER_END_OF_STREAM == state)) { /* * Yes. Drain the output buffer and stop playing. */ _DEBUG("End of stream reached, draining output buffer"); while((-1 == seek_to) && flac_ip.output->buffer_playing() && playback->playing == TRUE) { g_usleep(40000); } if (-1 == seek_to) { _DEBUG("Output buffer empty."); playback->playing = FALSE; } } } /* * Clean up a bit */ playback->playing = FALSE; _DEBUG("Closing audio device"); flac_ip.output->close_audio(); _DEBUG("Audio device closed"); free(play_buffer); if (FALSE == FLAC__stream_decoder_flush(main_decoder)) { _ERROR("Could not flush decoder state!"); } _LEAVE NULL; } /* --- */ void flac_play_file (InputPlayback* input) { VFSFile* fd; gint l; _ENTER; if (!plugin_initialized) { _ERROR("plugin not initialized!"); _LEAVE; } /* * Open the file */ if (NULL == (fd = aud_vfs_fopen(input->filename, "rb"))) { _ERROR("Could not open file for reading! (%s)", input->filename); _LEAVE; } if (FALSE == read_metadata(fd, main_decoder, main_info)) { _ERROR("Could not prepare file for playing!"); _LEAVE; } /* * Calculate the length */ if (0 == main_info->stream.samplerate) { _ERROR("Invalid sample rate for stream!"); l = -1; } else { l = (main_info->stream.samples / main_info->stream.samplerate) * 1000; } input->playing = TRUE; input->set_params(input, get_title(input->filename, main_info), l, -1, main_info->stream.samplerate, main_info->stream.channels); thread = g_thread_self(); input->set_pb_ready(input); flac_play_loop(input); _LEAVE; } /* --- */ void flac_stop(InputPlayback* input) { _ENTER; input->playing = FALSE; if (NULL != thread) { /* * Wait for the decoder thread to finish */ _DEBUG("Waiting for decoder thread to die..."); g_thread_join(thread); thread = NULL; _DEBUG("Decoder thread has finished"); } reset_info(main_info, TRUE); _LEAVE; } /* --- */ void flac_pause(InputPlayback* input, gshort p) { _ENTER; input->output->pause(p); _LEAVE; } /* --- */ void flac_seek(InputPlayback* input, gint time) { _ENTER; if (!input->playing) { _DEBUG("Can not seek while not playing"); _LEAVE; } _DEBUG("Requesting seek to %d", time); seek_to = time; while (-1 != seek_to) { g_usleep(10000); } _LEAVE; } /* --- */ void flac_get_song_info(gchar* filename, gchar** title, gint* length) { gint l; VFSFile* fd; _ENTER; _DEBUG("Testing file: %s", filename); /* * Open the file */ if (NULL == (fd = aud_vfs_fopen(filename, "rb"))) { _ERROR("Could not open file for reading! (%s)", filename); _LEAVE; } INFO_LOCK(test_info); if (FALSE == read_metadata(fd, test_decoder, test_info)) { _ERROR("Could not read file info!"); *length = -1; *title = g_strdup(""); reset_info(test_info, TRUE); INFO_UNLOCK(test_info); _LEAVE; } /* * Calculate the stream length (milliseconds) */ if (0 == test_info->stream.samplerate) { _ERROR("Invalid sample rate for stream!"); l = -1; } else { l = (test_info->stream.samples / test_info->stream.samplerate) * 1000; _DEBUG("Stream length: %d seconds", l/1000); } *length = l; *title = get_title(filename, test_info); reset_info(test_info, TRUE); INFO_UNLOCK(test_info); _LEAVE; } /* --- */ Tuple *flac_get_song_tuple(gchar* filename) { VFSFile *fd; Tuple *tuple; _ENTER; _DEBUG("Testing file: %s", filename); /* * Open the file */ if (NULL == (fd = aud_vfs_fopen(filename, "rb"))) { _ERROR("Could not open file for reading! (%s)", filename); _LEAVE NULL; } INFO_LOCK(test_info); if (FALSE == read_metadata(fd, test_decoder, test_info)) { _ERROR("Could not read metadata tuple for file <%s>", filename); reset_info(test_info, TRUE); INFO_UNLOCK(test_info); _LEAVE NULL; } tuple = get_tuple(filename, test_info); reset_info(test_info, TRUE); INFO_UNLOCK(test_info); _LEAVE tuple; } /* --- */ void flac_aboutbox(void) { static GtkWidget* about_window; gchar *about_text; if (about_window) { gtk_window_present(GTK_WINDOW(about_window)); return; } about_text = g_strjoin("", _("FLAC Audio Plugin "), _VERSION, _("\n\nOriginal code by\n" "Ralf Ertzinger <ralf@skytale.net>\n" "\n" "http://www.skytale.net/projects/bmp-flac2/"), NULL); about_window = audacious_info_dialog(_("About FLAC Audio Plugin"), about_text, _("OK"), FALSE, NULL, NULL); g_signal_connect(G_OBJECT(about_window), "destroy", G_CALLBACK(gtk_widget_destroyed), &about_window); g_free(about_text); }