/* * Sweep, a sound wave editor. * * Copyright (C) 2000 Conrad Parker * Copyright (C) 2002 CSIRO Australia * */ /* * This file adapted from "player.c" in the "mad MPEG audio decoder" * source distribution: * * Copyright (C) 2000-2001 Robert Leslie * * 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. */ #ifdef HAVE_CONFIG_H # include #endif #ifdef HAVE_MAD #include #include #include #include #include #include #include #include #include #include #include #include #include #define BUFFER_LEN 4096 #include #include #include #include #include #include #include #include #include #include "sample.h" #include "interface.h" #include "file_dialogs.h" #include "file_sndfile.h" #include "question_dialogs.h" #include "preferences.h" #include "print.h" #include "view.h" static gboolean file_is_mpeg_audio (const char * pathname) { int fd; /* FIXME * the mpeg frame marker can occur in non mpeg files so * one blind check for a marker can produce a false postive. * similarly, due to the metadata tags, the max * frame size may also not be a large enough window within * which to find a frame in a legal mpegfile. * * reimplement this check as a contextual verfification of * a sequence of frames, starting from somewhere in the middle * of a file. in the mean time, raise size of buf from 2048 to 8192 * to mitigate the chance of a false negative. (with an increased * change of a false positive as a result.) */ unsigned char buf[8192]; int n, i; fd = open (pathname, O_RDONLY); if (fd == -1) goto out_false; n = read (fd, buf, sizeof (buf)); if (n < 4) goto out_false; /* Check for MPEG frame marker */ for (i = 0; i < sizeof (buf)-1; i++) { if ((buf[i] & 0xff) == 0xff && (buf[i+1] & 0xe0) == 0xe0) { goto out_true; } } out_false: close (fd); return FALSE; out_true: close (fd); return TRUE; } /* * This is a private message structure. A generic pointer to this structure * is passed to each of the callback functions. Put here any data you need * to access from within the callbacks. */ struct mad_info { sw_sample * sample; /* file size */ size_t length; /* input buffer */ unsigned char const *start; unsigned long offset; unsigned long remaining; unsigned char * end_buffer; int eof; /* output */ sw_framecount_t nr_frames; }; /* * This is the input callback. The purpose of this callback is to (re)fill * the stream buffer which is to be decoded. In this example, an entire file * has been mapped into memory, so we just call mad_stream_buffer() with the * address and length of the mapping. When this callback is called a second * time, we are finished decoding. */ static enum mad_flow input(void *data, struct mad_stream *stream) { struct mad_info * info = data; unsigned long n; if (info->eof) return MAD_FLOW_STOP; if (stream->next_frame) { info->offset = stream->next_frame - info->start; info->remaining = info->length - info->offset; n = MIN (info->remaining, BUFFER_LEN); if (n == info->remaining) { info->end_buffer = g_malloc0 (info->remaining + MAD_BUFFER_GUARD); if (info->end_buffer == NULL) return MAD_FLOW_BREAK; info->eof = 1; memcpy (info->end_buffer, info->start + info->offset, info->remaining); mad_stream_buffer (stream, info->end_buffer, info->remaining + MAD_BUFFER_GUARD); info->offset += n; info->remaining -= n; return MAD_FLOW_CONTINUE; } } else { n = MIN (info->remaining, BUFFER_LEN); } mad_stream_buffer(stream, info->start + info->offset, n); info->offset += n; info->remaining -= n; return MAD_FLOW_CONTINUE; } /* * This is the output callback function. It is called after each frame of * MPEG audio data has been completely decoded. The purpose of this callback * is to output (or play) the decoded PCM audio. */ static enum mad_flow output(void *data, struct mad_header const *header, struct mad_pcm *pcm) { struct mad_info * info = data; sw_sample * sample = info->sample; sw_framecount_t data_start; sw_audio_t * d; int i, j; gint percent; gboolean active = TRUE; g_mutex_lock (sample->ops_mutex); if (sample->edit_state == SWEEP_EDIT_STATE_CANCEL) { active = FALSE; } else { sample->sounddata->format->channels = pcm->channels; sample->sounddata->format->rate = pcm->samplerate; data_start = info->nr_frames; info->nr_frames += pcm->length; if (info->nr_frames > sample->sounddata->nr_frames) { g_mutex_lock(sample->sounddata->data_mutex); sample->sounddata->data = g_realloc(sample->sounddata->data, frames_to_bytes (sample->sounddata->format, info->nr_frames)); g_mutex_unlock(sample->sounddata->data_mutex); } sample->sounddata->nr_frames = info->nr_frames; d = (sw_audio_t *)sample->sounddata->data; d = &d[data_start * pcm->channels]; for (i = 0; i < pcm->channels; i++) { for (j = 0; j < pcm->length; j++) { d[j*pcm->channels + i] = (sw_audio_t)mad_f_todouble(pcm->samples[i][j]); } } percent = (info->length - info->remaining) * 100 / info->length; sample_set_progress_percent (info->sample, percent); #ifdef DEBUG printf ("decoded %u samples, %d%% percent complete (%u / %u)\n", nsamples, percent, info->remaining, info->length); #endif } g_mutex_unlock (sample->ops_mutex); return (active ? MAD_FLOW_CONTINUE : MAD_FLOW_STOP); } /* * This is the error callback function. It is called whenever a decoding * error occurs. The error is indicated by stream->error; the list of * possible MAD_ERROR_* errors can be found in the mad.h (or * libmad/stream.h) header file. */ static enum mad_flow error(void *data, struct mad_stream *stream, struct mad_frame *frame) { struct mad_info * info = data; switch (stream->error) { case MAD_ERROR_BADDATAPTR: return MAD_FLOW_CONTINUE; case MAD_ERROR_LOSTSYNC: if (stream->next_frame) return MAD_FLOW_CONTINUE; /* else fall through */ default: fprintf(stderr, "decoding error 0x%04x (%s) at byte offset %ld\n", stream->error, mad_stream_errorstr(stream), stream->this_frame - info->start); break; } if (MAD_RECOVERABLE (stream->error)) return MAD_FLOW_CONTINUE; else return MAD_FLOW_BREAK; } static sw_sample * sample_load_mad_data (sw_op_instance * inst) { sw_sample * sample = inst->sample; int fd; void * fdm; struct stat statbuf; struct mad_decoder decoder; struct mad_info info; int result; fd = open (sample->pathname, O_RDONLY); if (fstat (fd, &statbuf) == -1 || statbuf.st_size == 0) return NULL; fdm = mmap (0, statbuf.st_size, PROT_READ, MAP_SHARED, fd, 0); if (fdm == MAP_FAILED) { perror (NULL); close (fd); return NULL; } #if defined (HAVE_MADVISE) madvise (fdm, statbuf.st_size, MADV_SEQUENTIAL); #endif info.sample = sample; info.length = statbuf.st_size; info.start = fdm; info.offset = 0; info.remaining = (unsigned long) statbuf.st_size; info.eof = 0; info.end_buffer = NULL; info.nr_frames = 0; mad_decoder_init (&decoder, &info, input, 0 /* header */, 0 /* filter */, output, error, 0 /* message */); result = mad_decoder_run (&decoder, MAD_DECODER_MODE_SYNC); mad_decoder_finish (&decoder); if (info.end_buffer != NULL) { g_free (info.end_buffer); } if (munmap (fdm, statbuf.st_size) == -1) { perror (NULL); } close (fd); stat (sample->pathname, &statbuf); sample->last_mtime = statbuf.st_mtime; sample->edit_ignore_mtime = FALSE; sample->modified = FALSE; sample_set_edit_state (sample, SWEEP_EDIT_STATE_DONE); return sample; } static sw_operation mad_load_op = { SWEEP_EDIT_MODE_FILTER, (SweepCallback)sample_load_mad_data, (SweepFunction)NULL, (SweepCallback)NULL, /* undo */ (SweepFunction)NULL, (SweepCallback)NULL, /* redo */ (SweepFunction)NULL }; static sw_sample * sample_load_mad_info (sw_sample * sample, char * pathname) { char buf[128]; gboolean isnew = (sample == NULL); sw_view * v; if (!file_is_mpeg_audio (pathname)) return NULL; /* Create the sample/sounddata, initially with length 0, to be grown * as the file is decoded */ if (sample == NULL) { /* Channels and rate will be set during decoding and are basically * irrelevent here. Set them to 2, 44100 assuming these are the most * likely values, in which case the file info displayed in the window * will not change suddenly */ sample = sample_new_empty(pathname, 2, 44100, 0); } else { int channels, rate; /* Set the channels and rate of the recreated sounddata to be the same * as the old one, as they are most likely the same after a reload */ channels = sample->sounddata->format->channels; rate = sample->sounddata->format->rate; sounddata_destroy (sample->sounddata); sample->sounddata = sounddata_new_empty (channels, rate, 0); } if(!sample) { /*close (fd);*/ return NULL; } sample->file_method = SWEEP_FILE_METHOD_MP3; /*sample->file_info = GINT_TO_POINTER(fd);*/ sample_bank_add(sample); if (isnew) { v = view_new_all (sample, 1.0); sample_add_view (sample, v); } else { trim_registered_ops (sample, 0); } g_snprintf (buf, sizeof (buf), _("Loading %s"), g_basename (sample->pathname)); schedule_operation (sample, buf, &mad_load_op, sample); return sample; } sw_sample * mad_sample_reload (sw_sample * sample) { if (sample == NULL) return NULL; return sample_load_mad_info (sample, sample->pathname); } sw_sample * mad_sample_load (char * pathname) { if (pathname == NULL) return NULL; return sample_load_mad_info (NULL, pathname); } #endif /* HAVE_MAD */